Computer Networks and the Internet

Transcription

1 C01 pp4 6/14/02 5:45 PM Page 1 Chapter 1 Computer Networks and the Internet Computer networking is one of the most exciting and important technological fields of our time. The Internet interconnects millions (and soon billions) of computers, providing a global communication, storage, and computation infrastructure. Moreover, the Internet is currently being integrated with mobile and wireless technology, ushering in an impressive array of new applications. Yes, computer networking has indeed come a long way since its infancy in the 1960s. But this is only the beginning a new generation of creative engineers and computer scientists will drive the Internet to yet unforeseen terrains. This book will provide today s students with the vehicles they need to journey to and explore the new lands in this exciting field. This first chapter presents an overview of computer networking and the Internet. Our goal here is to paint a broad-brush picture of computer networking, to see the forest through the trees. We ll cover a lot of ground in this introductory chapter and discuss a lot of pieces of a computer network, while not losing sight of the big picture. The chapter lays the groundwork for the rest of the book. It can also be used for a mini-course on computer networking. In this chapter, after introducing some basic terminology and concepts, we will first examine the edge of a computer network. We ll look at the end systems and network applications, and the transport services provided to these applications. We ll then explore the core of a computer network, examining the links and the switches that transport data, as well as the access networks and physical media that 1

2 C01 pp4 6/14/02 5:45 PM Page 2 2 CHAPTER 1 Computer Networks and the Internet connect end systems to the network core. We ll learn that the Internet is a network of networks, and we ll learn about how these networks connect with each other. After having completed this overview of the edge and core of a computer network, we ll take a broader view. We ll examine the causes of data-transfer delay and loss in a computer network, and provide simple quantitative models for end-toend delay, models that take into account transmission, propagation, and queuing delays. We ll then introduce some of the key architectural principles in computer networking, namely, protocol layering and service models. Finally, we ll close this chapter with a brief history of computer networking. 1.1 What Is the Internet? In this book we use the public Internet, a specific computer network, as our principal vehicle for discussing computer network protocols. But what is the Internet? We would like to be able to give you a one-sentence definition of the Internet, a definition that you can take home and share with your family and friends. Alas, the Internet is very complex, both in terms of its hardware and software components, as well as in the services it provides A Nuts-and-Bolts Description Instead of giving a one-sentence definition, let s try a more descriptive approach. There are a couple of ways to do this. One way is to describe the nuts and bolts of the Internet, that is, the basic hardware and software components that make up the Internet. Another way is to describe the Internet in terms of a networking infrastructure that provides services to distributed applications. Let s begin with the nuts-andbolts description, using Figure 1.1 to illustrate our discussion. The public Internet is a worldwide computer network, that is, a network that interconnects millions of computing devices throughout the world. Most of these computing devices are traditional desktop PCs, UNIX-based workstations, and so-called servers that store and transmit information such as Web pages and messages. Increasingly, nontraditional Internet end systems such as PDAs (Personal Digital Assistants), TVs, mobile computers, automobiles, and toasters are being connected to the Internet. (Toasters [Toasty 2002] are not the only rather unusual devices to have been hooked up to the Internet; see Internet Home Appliances [Appliance 2002].) In the Internet jargon, all of these devices are called hosts or end systems. As of January 2002 there were million end systems using the Internet; and this number continues to grow exponentially [ISC 2002]. End systems are connected together by communication links. We ll see in Section 1.4 that there are many types of communication links, which are made up of different types of physical media, including coaxial cable, copper wire, fiber optics,

3 C01 pp4 6/14/02 5:45 PM Page What Is the Internet? 3 Regional ISP Local ISP Company Network Key: Host (or end system) Server Mobile Router Modem Base station Satellite link Figure 1.1 Some pieces of the Internet and radio spectrum. Different links can transmit data at different rates. The link transmission rate is often called the bandwidth of the link, which is typically measured in bits/second. End systems are not usually directly attached to each other via a single communication link. Instead, they are indirectly connected to each other through intermediate switching devices known as routers. A router takes a chunk of information arriving on one of its incoming communication links and forwards that chunk of

4 C01 pp4 6/14/02 5:45 PM Page 4 4 CHAPTER 1 Computer Networks and the Internet information on one of its outgoing communication links. In the jargon of computer networking, the chunk of information is called a packet. The path that the packet takes from the sending end system, through a series of communication links and routers, to the receiving end system is known as a route or path through the network. Rather than provide a dedicated path between communicating end systems, the Internet uses a technique known as packet switching that allows multiple communicating end systems to share a path, or parts of a path, at the same time. The first packet-switched networks, created in the 1970s, are the earliest ancestors of today s Internet. End systems access the Internet through Internet Service Providers (ISPs),including residential ISPs such as AOL or MSN, university ISPs such as Stanford University, and corporate ISPs such as Ford Motor Company. Each ISP is a network of routers and communication links. The different ISPs provide a variety of different types of network access to the end systems, including 56 Kbps dial-up modem access, residential broadband access such as cable modem or DSL, high-speed LAN access, and wireless access. ISPs also provide Internet access to content providers, connecting Web sites directly to the Internet. To allow communication among Internet users and to allow users to access worldwide Internet content, these lower-tier ISPs are interconnected through national and international upper-tier ISPs, such as the UUNet and Sprint. An upper-tier ISP consists of high-speed routers interconnected with high-speed fiber-optic links. Each ISP network, whether upper-tier or lower-tier, is managed independently, runs the IP protocol (see below), and conforms to certain naming and address conventions. We will examine ISPs and their interconnection more closely in Section 1.5. End systems, routers, and other pieces of the Internet, run protocols that control the sending and receiving of information within the Internet. TCP (the Transmission Control Protocol) and IP (the Internet Protocol) are two of the most important protocols in the Internet. The IP protocol specifies the format of the packets that are sent and received among routers and end systems. The Internet s principal protocols are collectively known as TCP/IP. We begin looking into protocols in this introductory chapter. But that s just a start much of this book is concerned with computer network protocols! The public Internet (that is, the global network of networks discussed above) is the network that one typically refers to as the Internet. There are also many private networks, such as many corporate and government networks, whose hosts cannot exchange messages with hosts outside of the private network (unless the messages pass through so-called firewalls, which restrict the flow of messages to and from the network). These private networks are often referred to as intranets, as they use the same types of hosts, routers, links, and protocols as the public Internet. At the technical and developmental level, the Internet is made possible through creation, testing, and implementation of Internet standards. These standards are developed by the Internet Engineering Task Force (IETF)[IETF 2002]. The IETF standards documents are called RFCs (request for comments). RFCs started out as general requests for comments (hence the name) to resolve architecture problems

5 C01 pp4 6/14/02 5:45 PM Page What Is the Internet? 5 that faced the precursor to the Internet. RFCs, though not formally standards, have evolved to the point where they are cited as such. RFCs tend to be quite technical and detailed. They define protocols such as TCP, IP, HTTP (for the Web), and SMTP (for open-standards ). There are more than 3,000 different RFCs A Service Description The preceding discussion has identified many of the pieces that make up the Internet. Let s now leave the nuts-and-bolts description and take a service-oriented view. The Internet allows distributed applications running on its end systems to exchange data with each other. These applications include remote login, electronic mail, Web surfing, instant messaging, audio and video streaming, Internet telephony, distributed games, peer-to-peer (P2P) file sharing, and much, much more. It is worth emphasizing that the Web is not a separate network but rather just one of many distributed applications that use the communication services provided by the Internet. The Internet provides two services to its distributed applications: a connectionoriented reliable service and a connectionless unreliable service. Loosely speaking, the connection-oriented reliable service guarantees that data transmitted from a sender to a receiver will eventually be delivered to the receiver in order and in its entirety. The connectionless unreliable service does not make any guarantees about eventual delivery. Typically, a distributed application makes use of one or the other (but not both) of these two services. Currently, the Internet does not provide a service that makes promises about how long it will take to deliver the data from sender to receiver. And except for increasing your access bandwidth to your Internet service provider, you currently cannot obtain better service (for example, bounded delays) by paying more a state of affairs that some (particularly Americans!) find odd. We ll take a look at state-ofthe-art Internet research that is aimed at changing this situation in Chapter 6. This second description of the Internet that is, in terms of the services it provides to distributed applications is a nontraditional, but important, one. Increasingly, advances in the nuts-and-bolts components of the Internet are being driven by the needs of new applications. So it s important to keep in mind that the Internet is an infrastructure in which new applications are being constantly invented and deployed. We have just given two descriptions of the Internet, one in terms of its hardware and software components, the other in terms of the services it provides to distributed applications. But perhaps you are still confused as to what the Internet is. What are packet switching, TCP/IP, and connection-oriented service? What are routers? What kinds of communication links are present in the Internet? What is a distributed application? If you feel a bit overwhelmed by all of this now, don t worry the

6 C01 pp4 6/14/02 5:45 PM Page 6 Got the time? 6 CHAPTER 1 Computer Networks and the Internet purpose of this book is to introduce you to both the nuts and bolts of the Internet, as well as the principles that govern how and why it works. We will explain these important terms and questions in the subsequent sections and chapters What Is a Protocol? Now that we ve got a bit of a feel for what the Internet is, let s consider another important buzzword in computer networking: protocol. What is a protocol? What does a protocol do? How would you recognize a protocol if you met one? A Human Analogy It is probably easiest to understand the notion of a computer network protocol by first considering some human analogies, since we humans execute protocols all of the time. Consider what you do when you want to ask someone for the time of day. A typical exchange is shown in Figure 1.2. Human protocol (or good manners, at least) Hi TCP connection request Hi TCP connection reply GET 2:00 <file> Time Time Time Time Figure 1.2 A human protocol and a computer network protocol

7 C01 pp4 6/14/02 5:45 PM Page What Is the Internet? 7 dictates that one first offer a greeting (the first Hi in Figure 1.2) to initiate communication with someone else. The typical response to a Hi is a returned Hi message. Implicitly, one then takes a cordial Hi response as an indication that one can proceed and ask for the time of day. A different response to the initial Hi (such as Don t bother me! or I don t speak English, or some unprintable reply) might indicate an unwillingness or inability to communicate. In this case, the human protocol would be not to ask for the time of day. Sometimes one gets no response at all to a question, in which case one typically gives up asking that person for the time. Note that in our human protocol, there are specific messages we send, and specific actions we take in response to the received reply messages or other events (such as no reply within some given amount of time). Clearly, transmitted and received messages, and actions taken when these messages are sent or received or other events occur, play a central role in a human protocol. If people run different protocols (for example, if one person has manners but the other does not, or if one understands the concept of time and the other does not) the protocols do not interoperate and no useful work can be accomplished. The same is true in networking it takes two (or more) communicating entities running the same protocol in order to accomplish a task. Let s consider a second human analogy. Suppose you re in a college class (a computer networking class, for example!). The teacher is droning on about protocols and you re confused. The teacher stops to ask, Are there any questions? (a message that is transmitted to, and received by, all students who are not sleeping). You raise your hand (transmitting an implicit message to the teacher). Your teacher acknowledges you with a smile, saying Yes... (a transmitted message encouraging you to ask your question teachers love to be asked questions), and you then ask your question (that is, transmit your message to your teacher). Your teacher hears your question (receives your question message) and answers (transmits a reply to you). Once again, we see that the transmission and receipt of messages, and a set of conventional actions taken when these messages are sent and received, are at the heart of this question-and-answer protocol. Network Protocols A network protocol is similar to a human protocol, except that the entities exchanging messages and taking actions are hardware or software components of some device (for example, computer, router, or other network-capable device). All activity in the Internet that involves two or more communicating remote entities is governed by a protocol. For example, protocols in routers determine a packet s path from source to destination; hardware-implemented protocols in the network interface cards of two physically connected computers control the flow of bits on the wire between the two network interface cards; congestion-control protocols in end systems control the rate at which packets are transmitted between sender and receiver. Protocols are running everywhere in the Internet, and consequently much of this book is about computer network protocols.

8 C01 pp4 6/14/02 5:45 PM Page 8 8 CHAPTER 1 Computer Networks and the Internet As an example of a computer network protocol with which you are probably familiar, consider what happens when you make a request to a Web server, that is, when you type in the URL of a Web page into your Web browser. The scenario is illustrated in the right half of Figure 1.2. First, your computer will send a connection request message to the Web server and wait for a reply. The Web server will eventually receive your connection request message and return a connection reply message. Knowing that it is now OK to request the Web document, your computer then sends the name of the Web page it wants to fetch from that Web server in a GET message. Finally, the Web server returns the Web page (file) to your computer. Given the human and networking examples above, the exchange of messages and the actions taken when these messages are sent and received are the key defining elements of a protocol: A protocol defines the format and the order of messages exchanged between two or more communicating entities, as well as the actions taken on the transmission and/or receipt of a message or other event. The Internet, and computer networks in general, make extensive use of protocols. Different protocols are used to accomplish different communication tasks. As you read through this book, you will learn that some protocols are simple and straightforward, while others are complex and intellectually deep. Mastering the field of computer networking is equivalent to understanding the what, why, and how of networking protocols Some Good Hyperlinks As every Internet researcher knows, some of the best and most accurate information about the Internet and its protocols is not in hard-copy books, journals, or magazines. Some of the best stuff about the Internet is in the Internet itself! Of course, there s really too much material to sift through, and sometimes the gems are few and far between. Below, we list a few generally excellent Web sites for network- and Internet-related material. Throughout the book, we will also present links to relevant, high-quality URLs that provide background, original, or advanced material related to the particular topic under study. Here is a set of key links that you may want to consult as you proceed through this book: Internet Engineering Task Force (IETF), The IETF is an open international community concerned with the development and operation of the Internet and its architecture. The IETF was formally established by the Internet Architecture Board (IAB), in The IETF meets three times a year; much of its ongoing work is conducted via mailing lists by working groups. The IETF is administered by the Internet Society, whose Web site contains lots of high-quality, Internet-related material.

9 C01 pp4 6/14/02 5:45 PM Page The Network Edge 9 The World Wide Web Consortium (W3C), The W3C was founded in 1994 to develop common protocols for the evolution of the World Wide Web. This is an outstanding site with fascinating information on emerging Web technologies, protocols, and standards. The Association for Computing Machinery (ACM), and the Institute of Electrical and Electronics Engineers (IEEE), These are the two main international professional societies that have technical conferences, magazines, and journals in the networking area. The ACM Special Interest Group in Data Communications (SIGCOMM), sigcomm, the IEEE Communications Society, and the IEEE Computer Society, are the groups within these bodies whose efforts are most closely related to networking. Computer Networking: A Top-Down Approach Featuring the Internet (that is, the Web site for this textbook!), You ll find a wealth of resources at the Web site, including hyperlinks to relevant Web pages, Java applets illustrating networking concepts, homework problems with answers, programming projects, streaming audio lectures coupled to slides, and much more. 1.2 The Network Edge In the previous sections we presented a high-level overview of the Internet and networking protocols. We are now going to delve a bit more deeply into the components of a computer network (and the Internet, in particular). We begin in this section at the edge of a network and look at the components with which we are most familiar namely, the computers that we use on a daily basis. In the next section we will move from the network edge to the network core and examine switching and routing in computer networks. Then in Section 1.4 we will discuss the actual physical links that carry the signals sent between computers and switches End Systems, Clients, and Servers In computer networking jargon, the computers connected to the Internet are often referred to as end systems. They are referred to as end systems because they sit at the edge of the Internet, as shown in Figure 1.3. The Internet s end systems include many different types of computers. End users directly interface with some of these computers, including desktop computers (desktop PCs, Macs, and UNIX-based workstations) and mobile computers (portable computers and PDAs with wireless Internet connections). The Internet s end systems also include computers with which users do not directly interface, such as Web servers and servers. Furthermore, an increasing number of alternative devices, such as thin clients and household

10 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet Regional ISP Local ISP Company Network Figure 1.3 End-system interaction appliances [Thinplanet 2002], Web TVs and set top boxes [Nesbitt 2002], and digital cameras are being attached to the Internet as end systems. For interesting discussions of the future of Internet appliances see [Manelli 2001; Appliance 2001; Dertouzos 2001; Odlyzko 1999]. End systems are also referred to as hosts because they host (that is, run) application programs such as a Web browser program, a Web server program, an reader program, or an server program. Throughout this book we will use the terms hosts and end systems interchangeably; that is, host = end system. Hosts are sometimes further divided into two categories: clients and servers. Informally, clients tend to be desktop and mobile PCs, PDAs, and so on, whereas servers tend to be more powerful machines hosting servers such as Web servers and mail servers.

11 C01 pp4 6/14/02 5:45 PM Page The Network Edge 11 In the context of networking software, there is another definition of a client and server, a definition that we ll refer to throughout this book. A client program is a program running on one end system that requests and receives a service from a server program running on another end system. Studied in detail in Chapter 2, this client/server model is undoubtedly the most prevalent structure for Internet applications. The Web, , file transfer, remote login (for example, Telnet), newsgroups, and many other popular applications adopt the client/server model. Since a client program typically runs on one computer and the server program runs on another computer, client/server Internet applications are, by definition, distributed applications. The client program and the server program interact with each other by sending each other messages over the Internet. At this level of abstraction, the routers, links and other nuts and bolts of the Internet serve as a black box that transfers messages between the distributed, communicating components of an Internet application. This is the level of abstraction depicted in Figure 1.3. Not all Internet applications today consist of pure client programs interacting with pure server programs. For example, with the popular peerto-peer file sharing applications (Napster, Gnutella, KaZaA, and so on), the peer-to-peer application in the user s end system acts as both a client program and a server program. The program running in a peer (that is, a user s machine) acts as a client when it requests a file from another peer; and the program acts as a server when it sends a file to another peer. Case History SEARCH FOR EXTRATERRESTRIAL LIFE One of the niftiest applications of the Internet is the SETI@home project [SETI@home 2002], a scientific experiment that uses Internet-connected computers in the Search for Extraterrestrial Intelligence (SETI). Anyone can participate by running a free client program that downloads and analyzes radio telescope data. The goal of the project is to find, in the radio telescope data, signals that were created by extraterrestrial life. SETI@home searches for signs of intelligent life by analyzing radio data that are collected from Arecibo, the largest radio telescope in the world, located in the hills of northern Puerto Rico. Massive quantities of radio data are collected on tapes and sent to Berkeley every week. At Berkeley, the data is divided into work units of about 300 Kbytes, which are stored in a central SETI@home server. To participate in this project, an Internet user (for example, you!) first downloads a client program from SETI@home that runs in the background on a host computer (for example, your PC). The client program then sets up a TCP connection to the central server, obtains a work unit, and closes the connection. Once a host has obtained a work unit, it processes the data mostly FFT (Fast Fourier Transform) calculations which may take from an hour to several days, depending on the power and usage of the host. When the calculations are finished, the client program reconnects to the central server, sends back the results, and gets a new work unit. Today, over 3 million users from more than 200 countries have downloaded and run the client program. In a typical day, the hosts together perform over 20 trillion floating-point operations per second, which is faster than the largest of the supercomputers. And the SETI@home project is only the tip of the iceberg for peerbased scientific computing. If 10 percent of the approximately one billion Internetconnected hosts participate in peer-based computing projects, there will be enough computing power for 100 projects the size of SETI@home [Anderson 2001].

12 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet Connectionless and Connection-Oriented Service End systems use the Internet to communicate with each other. Specifically, endsystem programs use the services of the Internet to send messages to each other. The links, routers, and other pieces of the Internet provide the means to transport these messages between the end-system programs. But what are the characteristics of the communication services that the Internet provides to its end systems? TCP/IP networks, and in particular the Internet, provide two types of services to end-system applications: connectionless service and connection-oriented service. A developer creating an Internet application (for example, an application, a file transfer application, a Web application, or an Internet phone application) must design the application to use one of these two services. We now briefly describe these two services. (We ll discuss these two services in much more detail in Chapter 3, which covers transport layer protocols.) Connection-Oriented Service When an application uses the connection-oriented service, the client program and the server program (residing in different end systems) send control packets to each other before sending packets with real data (such as messages). This socalled handshaking procedure alerts the client and server, allowing them to prepare for an onslaught of packets. Once the handshaking procedure is finished, a connection is said to be established between the two end systems. It is interesting to note that this initial handshaking procedure is similar to the protocol used in human interaction. The exchange of Hi s we saw in Figure 1.2 is an example of a human handshaking protocol (even though handshaking is not literally taking place between the two people). For the Web interaction also shown in Figure 1.2, the first two messages exchanged are also handshaking messages. The subsequent two messages the GET message and the response message containing the file include real data and are sent only after the connection has been established. Why the terminology connection-oriented service and not just connection service? This terminology is due to the fact that the end systems are connected in a very loose manner. In particular, only the end systems themselves are aware of this connection; the packet switches (that is, routers) within the Internet are completely oblivious to the connection. Indeed, a connection in the Internet consists of nothing more than allocated buffers and state variables in the end systems; the intervening packet switches do not maintain any connection-state information. The Internet s connection-oriented service comes bundled with several other services, including reliable data transfer, flow control, and congestion control. By reliable data transfer, we mean that an application can rely on the connection to deliver all of its data without error and in the proper order. Reliability in the Internet is achieved through the use of acknowledgments and retransmissions. To get a preliminary feel for how the Internet implements the reliable transport service, consider an application that has established a connection between end systems A and B.

13 C01 pp4 6/14/02 5:45 PM Page The Network Edge 13 When end system B receives a packet from A, it sends an acknowledgment; when end system A receives the acknowledgment, it knows that the corresponding packet has definitely been received. When end system A doesn t receive an acknowledgment, it assumes the packet it sent wasn t received by B and thus retransmits the packet. Flow control makes sure that neither side of a connection overwhelms the other side by sending too many packets too fast. Indeed, there is a risk that the receiver may not be able to keep up with the rate at which the sender is sending packets. The flow-control service forces the sending end system to reduce its rate whenever there is such a risk. We ll see in Chapter 3 that the Internet implements the flow-control service by using sender and receiver buffers in the communicating end systems. The Internet s congestion-control service helps prevent the Internet from entering a state of gridlock. When a router becomes congested, its buffers can overflow and packet loss can occur. In such circumstances, if every pair of communicating end systems continues to pump packets into the network as fast as they can, gridlock sets in and few packets are delivered to their destinations. The Internet avoids this problem by forcing end systems to decrease the rate at which they send packets into the network during periods of congestion. End systems are alerted to the existence of severe congestion when they stop receiving acknowledgments for the packets they have sent. We emphasize here that although the Internet s connection-oriented service comes bundled with reliable data transfer, flow control, and congestion control, these three features are by no means essential components of a connection-oriented service. A different type of computer network may provide a connection-oriented service to its applications without bundling in one or more of these features. Indeed, any protocol that performs handshaking between the communicating entities before transferring data is a connection-oriented service [Iren 1999]. The Internet s connection-oriented service has a name TCP (Transmission Control Protocol); the initial version of the TCP protocol is defined in the Internet Request for Comments RFC 793 [RFC 793]. The services that TCP provides to an application include reliable transport, flow control, and congestion control. It is important to note that an application need only care about the services that are provided; it need not worry about how TCP actually implements reliability, flow control, or congestion control. We, of course, are very interested in how TCP implements these services, and we shall cover these topics in detail in Chapter 3. Connectionless Service There is no handshaking with the Internet s connectionless service. When one side of an application wants to send packets to the other side of the application, the sending program simply sends the packets. Since there is no handshaking procedure prior to data packet transmission, data can be delivered sooner. But there is no reliable data transfer either, so a source never knows for sure which packets have arrived at the destination. Moreover, the Internet s connectionless service makes no provision for flow control or congestion control. The Internet s connectionless

14 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet service is called UDP (User Datagram Protocol); UDP is defined in the Internet Request for Comments RFC 768. Most of the more familiar Internet applications use TCP, the Internet s connectionoriented service. These applications include Telnet (for remote login), SMTP (for electronic mail), FTP (for file transfer), and HTTP (for the Web). Nevertheless, UDP, the Internet s connectionless service, is used by many applications, including many of the emerging multimedia applications, such as Internet phone and video conferencing. 1.3 The Network Core Having examined the end systems and end-end transport service model of the Internet, let us now delve more deeply into the inside of the network. In this section we study the network core the mesh of routers that interconnect the Internet s end systems. Figure 1.4 highlights the network core with thick, shaded lines Circuit Switching and Packet Switching There are two fundamental approaches toward building a network core: circuit switching and packet switching. In circuit-switched networks, the resources needed along a path (buffers, link bandwidth) to provide for communication between the end systems are reserved for the duration of the communication session. In packet-switched networks, these resources are not reserved; a session s messages use the resources on demand, and as a consequence, may have to wait (that is, queue) for access to a communication link. As a simple analogy, consider two restaurants one that requires reservations and another that neither requires reservations nor accepts them. For the restaurant that requires reservations, we have to go through the hassle of first calling before we leave home. But when we arrive at the restaurant we can, in principle, immediately communicate with the waiter and order our meal. For the restaurant that does not require reservations, we don t need to bother to reserve a table. But when we arrive at the restaurant, we may have to wait for a table before we can communicate with the waiter. The ubiquitous telephone networks are examples of circuit-switched networks. Consider what happens when one person wants to send information (voice or facsimile) to another over a telephone network. Before the sender can send the information, the network must first establish a connection between the sender and the receiver. In contrast with the TCP connection that we discussed in the previous section, this is a bona fide connection for which the switches on the path between the sender and receiver maintain connection state for that connection. In the jargon of telephony, this connection is called a circuit. When the network establishes the circuit, it also reserves a constant transmission rate in the network s links for the duration of the connection. Since bandwidth has been reserved for this sender-to-receiver connection, the sender can transfer the data to the receiver at the guaranteed constant rate.

15 C01 pp4 6/14/02 5:45 PM Page The Network Core 15 Regional ISP Local ISP Company Network Figure 1.4 The network core Today s Internet is a quintessential packet-switched network. Consider what happens when one host wants to send a packet to another host over the Internet. As with circuit switching, the packet is transmitted over a series of communication links. But with packet switching, the packet is sent into the network without reserving any bandwidth whatsoever. If one of the links is congested because other packets need to be transmitted over the link at the same time, then our packet will have to wait in a buffer at the sending side of the transmission link, and suffer a delay. The Internet makes its best effort to deliver packets in a timely manner, but it does not make any guarantees. Not all telecommunication networks can be neatly classified as pure circuitswitched networks or pure packet-switched networks. For example, for networks based on the Asynchronous Transfer Mode (ATM) technology (a topic we consider

16 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet in Chapter 5), a connection can make a reservation and yet its messages may still wait for congested resources! Nevertheless, this fundamental classification into packet- and circuit-switched networks is an excellent starting point in understanding telecommunication network technology. Circuit Switching This book is about computer networks, the Internet, and packet switching, not about telephone networks and circuit switching. Nevertheless, it is important to understand why the Internet and other computer networks use packet switching rather than the more traditional circuit-switching technology used in the telephone networks. For this reason, we now give a brief overview of circuit switching. Figure 1.5 illustrates a circuit-switched network. In this network, the four circuit switches are interconnected by four links. Each of these links has n circuits, so that each link can support n simultaneous connections. The hosts (for example, PCs Host A Each link consists of n circuits (TDM or FDM) End-to-end connection between Hosts A and B, using one circuit in each of the links Host B Key: Host Circuit switch Figure 1.5 A simple circuit-switched network consisting of four switches and four links

17 C01 pp4 6/14/02 5:45 PM Page The Network Core 17 and workstations) are each directly connected to one of the switches. When two hosts want to communicate, the network establishes a dedicated end-to-end connection between two hosts. (Conference calls between more than two devices are, of course, also possible. But to keep things simple, let s suppose for now that there are only two hosts for each connection.) Thus, in order for host A to send messages to host B, the network must first reserve one circuit on each of two links. Because each link has n circuits, for each link used by the end-to-end connection, the connection gets the fraction 1/n of the link s bandwidth for the duration of the connection. Multiplexing in Circuit-Switched Networks A circuit in a link is implemented with either frequency-division multiplexing (FDM) or time-division multiplexing (TDM). With FDM, the frequency spectrum of a link is shared among the connections established across the link. Specifically, the link dedicates a frequency band to each connection for the duration of the connection. In telephone networks, this frequency band typically has a width of 4 khz (that is, 4,000 Hertz or 4,000 cycles per second). The width of the band is called, not surprisingly, the bandwidth. FM radio stations also use FDM to share the microwave frequency spectrum. The trend in modern telephony is to replace FDM with TDM. Most links in most telephone systems in developed countries currently employ TDM. For a TDM link, time is divided into frames of fixed duration, and each frame is divided into a fixed number of time slots. When the network establishes a connection across a link, the network dedicates one time slot in every frame to the connection. These slots are dedicated for the sole use of that connection, with a time slot available for use (in every frame) to transmit the connection s data. Figure 1.6 illustrates FDM and TDM for a specific network link supporting up to four circuits. For FDM, the frequency domain is segmented into four bands, each of bandwidth 4 khz. For TDM, the time domain is segmented into frames, with four time slots in each frame; each circuit is assigned the same dedicated slot in the revolving TDM frames. For TDM, the transmission rate of a circuit is equal to the frame rate multiplied by the number of bits in a slot. For example, if the link transmits 8,000 frames per second and each slot consists of eight bits, then the transmission rate of a circuit is 64 Kbps. Proponents of packet switching have always argued that circuit switching is wasteful because the dedicated circuits are idle during silent periods. For example, when one person in a telephone call stops talking, the idle network resources (frequency bands or slots in the links along the connection s route) cannot be used by other ongoing connections. As another example of how these resources can be underutilized, consider a radiologist who uses a circuit-switched network to remotely access a series of x-rays. The radiologist sets up a connection, requests an image, contemplates the image, and then requests a new image. Network resources are wasted during the radiologist s contemplation periods. Proponents of packet switching also enjoy

18 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet 4KHz FDM Link Frequency 4KHz TDM Key: Slot Frame Time 2 All slots labeled 2 are dedicated to a specific sender-receiver pair. Figure 1.6 With FDM, each circuit continuously gets a fraction of the bandwidth. With TDM, each circuit gets all of the bandwidth periodically during brief intervals of time (that is, during slots). pointing out that establishing end-to-end circuits and reserving end-to-end bandwidth is complicated and requires complex signaling software to coordinate the operation of the switches along the end-to-end path. Before we finish our discussion of circuit switching, let s work through a numerical example that should shed further insight on the topic. Let us consider how long it takes to send a file of 640,000 bits from host A to host B over a circuit-switched network. Suppose that all links in the network use TDM with 24 slots and have a bit rate of Mbps. Also suppose that it takes 500 msec to establish an end-to-end circuit before host A can begin to transmit the file. How long does it take to send the file? Each circuit has a transmission rate of (1.536 Mbps)/24 = 64 Kbps, so it takes (640,000 bits)/(64 Kbps) = 10 seconds to transmit the file. To this 10 seconds we add the circuit establishment time, giving 10.5 seconds to send the file. Note that the transmission time is independent of the number of links: The transmission time would be 10 seconds if the end-to-end circuit passed through one link or 100 links. (The actual end-to-end delay also includes a propagation delay; see Section 1.6.) Packet Switching We saw in Section 1.1 that applications exchange messages in accomplishing their task. Messages can contain anything the protocol designer wants. Messages may

19 C01 pp4 6/14/02 5:45 PM Page The Network Core 19 perform a control function (for example, the Hi messages in our handshaking example) or can contain data, such as an message, a JPEG image, or an MP3 audio file. In modern computer networks, the source breaks long messages into smaller chunks of data known as packets. Between source and destination, each of these packets travels through communication links and packet switches (also known as routers). Packets are transmitted over each communication link at a rate equal to the full transmission rate of the link. Most packet switches use store-andforward transmission at the inputs to the links. Store-and-forward transmission means that the switch must receive the entire packet before it can begin to transmit the first bit of the packet onto the outbound link. Thus store-and-forward packet switches introduce a store-and-forward delay at the input to each link along the packet s route. This delay is proportional to the packet s length in bits. In particular, if a packet consists of L bits, and the packet is to be forwarded onto an outbound link of R bps, then the store-and-forward delay at the switch is L/R seconds. Each router has multiple links attached to it. For each attached link, the router has an output buffer (also called an output queue), which stores packets that the router is about to send into that link. The output buffers play a key role in packet switching. If an arriving packet needs to be transmitted across a link but finds the link busy with the transmission of another packet, the arriving packet must wait in the output buffer. Thus, in addition to the store-and-forward delays, packets suffer output buffer queuing delays. These delays are variable and depend on the level of congestion in the network. Since the amount of buffer space is finite, an arriving packet may find that the buffer is completely filled with other packets waiting for transmission. In this case, packet loss will occur either the arriving packet or one of the already-queued packets will be dropped. Returning to our restaurant analogy from earlier in this section, the queuing delay is analogous to the amount of time you spend waiting at the restaurant s bar for a table to become free. Packet loss is analogous to being told by the waiter that you must leave the premises because there are already too many other people waiting at the bar for a table. Figure 1.7 illustrates a simple packet-switched network. In this and subsequent figures, packets are represented by three-dimensional slabs. The width of a slab represents the packet s length. In this figure, all packets have the same width and hence the same length. Suppose hosts A and B are sending packets to host E. Hosts A and B first send their packets along 10 Mbps Ethernet links to the first packet switch. The packet switch directs these packets to the 1.5 Mbps link. If there is congestion at this link, the packets will queue in the link s output buffer before being transmitted onto the link. Consider now how host A and host B packets are transmitted onto this link. As shown in Figure 1.7, the sequence of A and B packets does not follow any fixed pattern; instead the ordering is random because packets are sent whenever they happen to be present at the link. Because of this random ordering, we often say that packet switching employs statistical multiplexing. Statistical multiplexing sharply contrasts with time-division multiplexing (TDM) in circuit switching, for which each host gets the same slot in a revolving TDM frame.

20 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet A 10 Mbps Ethernet Statistical multiplexing C 1.5 Mbps Queue of packets waiting for output link B Key: D E Packets Figure 1.7 Packet switching Let s now consider how long it takes to send a packet of L bits from one host to another host across a packet-switched network. Let s suppose that there are Q links between the two hosts, each of rate R bps. Assume that queuing delays and end-toend propagation delays are negligible and that there is no connection establishment. The packet must first be transmitted onto the first link emanating from host A; this takes L/R seconds. It must then be transmitted on each of the Q 1 remaining links; that is, it must be stored and forwarded Q 1 times. Thus the total delay is QL/R. Packet Switching Versus Circuit Switching Having described circuit switching and packet switching, let us compare the two. Opponents of packet switching have often argued that packet switching is not suitable for real-time services (for example, telephone calls and video conference calls) because of its variable and unpredictable end-to-end delays (due primarily to variable and unpredictable queuing delays). Proponents of packet switching argue that (1) it offers better sharing of bandwidth than circuit switching and (2) it is simpler, more efficient, and less costly to implement than circuit switching. Generally speaking, people who do not like to hassle with restaurant reservations prefer packet switching to circuit switching. Why is packet switching more efficient? Let us look at a simple example. Suppose users share a 1 Mbps link. Also suppose that each user alternates between periods of activity, when the user generates data at a constant rate of 100 Kbps, and periods of inactivity (when the user generates no data). Suppose further that the user is active only 10 percent of the time (and is idle drinking coffee during the remain-

21 C01 pp4 6/14/02 5:45 PM Page The Network Core 21 ing 90 percent of the time). With circuit switching, 100 Kbps must be reserved for each user at all times. Thus, the link can support only 10 ( = 1Mbps/100 Kbps) simultaneous users. With packet switching, the probability that a specific user is active is 0.1 (that is, 10 percent). If there are 35 users, the probability that there are 11 or more simultaneously active users is approximately (Problem 10 outlines how this probability is obtained.) When there are 10 or fewer simultaneously active users (which happens with probability ), the aggregate arrival rate of data is less than or equal to 1 Mbps, the output rate of the link. Thus, when there are 10 or fewer active users, users packets flow through the link essentially without delay, as is the case with circuit switching. When there are more than 10 simultaneously active users, then the aggregate arrival rate of packets exceeds the output capacity of the link, and the output queue will begin to grow. (It continues to grow until the aggregate input rate falls back below 1 Mbps, at which point the queue will begin to diminish in length.) Because the probability of having more than 10 simultaneously active users is minuscule in this example, packet switching almost always has the same delay performance as circuit switching, but does so while allowing for more than three times the number of users. Although packet switching and circuit switching are both prevalent in today s telecommunication networks, the trend is certainly in the direction of packet switching. Even many of today s circuit-switched telephone networks are slowly migrating toward packet switching. In particular, telephone networks often use packet switching for the expensive overseas portion of a telephone call. Message Segmenting In a modern packet-switched network, the source host segments long, applicationlayer messages into smaller packets and sends the smaller packets into the network; the receiver later reassembles the packets back into the original messages. But why bother to segment the messages into packets in the first place, only to have to reassemble packets into messages? Doesn t this place an additional and unnecessary burden on the source and destination? Although segmentation and reassembly do complicate the design of the source and receiver, researchers and network designers concluded in the early days of packet switching that the advantages of segmentation greatly compensate for its complexity. Before discussing some of these advantages, we need to introduce some terminology. We say that a packet-switched network performs message switching if the sources do not segment messages (that is, they send a message into the network as a whole). Thus message switching is a specific kind of packet switching, in which the packets traversing the network are themselves entire application messages. Figure 1.8 illustrates message switching in a route consisting of two packet switches and three links. With message switching, the message stays intact as it traverses the network. Because the switches are store-and-forward packet switches, a packet switch must receive the entire message before it can begin to forward the

22 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet Message Source Packet switch Packet switch Destination Figure 1.8 Message sent from source to destination without message segmentation Packet Source Packet switch Packet switch Destination Figure 1.9 Message segmented into packets before being sent from source to destination message on an outbound link. Note the long width of the message, illustrating the fact that a long message that has not been segmented into packets. Figure 1.9 illustrates packet switching (with segmented messages) for the same network. In this example, the original message has been divided into four distinct packets. In Figure 1.9, the first packet has arrived at the destination, the second and third packets are in transit in the network, and the last packet is still in the source. Again, because the switches are store-and-forward packet switches, a packet switch must receive an entire packet before it can begin to forward the packet on an outbound link. When the message is segmented into packets, the network is said to pipeline message transmission that is, portions of the message are transmitted in parallel by the source and the two packet switches. (The term pipelining comes from the computer architecture and parallel processing literature, but also applies here to computer networks.) One major advantage of packet switching (with segmented messages) is that it achieves end-to-end delays that are typically much smaller than the delays associated with message switching. We illustrate this point with the following simple example. Consider a message that is bits long. Suppose that between source and destination there are two packet switches and three links, and that each link has a transmission rate of 1.5 Mbps. Assuming there is no congestion in the network, how much time is required to move the message from source to destination with message switching? It takes the source 5 seconds ( = bits / 1.5 Mbps) to move the message from the source to the first switch. Because the switches use store-and-forward transmission, the first switch cannot begin to transmit any bits in the message onto the link until this first switch has received the entire message. Once the first switch has received the entire message, it takes 5 seconds to move the message from the first switch to the second switch. Thus it takes 10 seconds to move

23 C01 pp4 6/14/02 5:45 PM Page The Network Core 23 Source Destination Time (in seconds) Key: Message Figure 1.10 Timing of message transfer without message segmentation the message from the source to the second switch. Following this logic we see that a total of 15 seconds is needed to move the message from source to destination. These delays are illustrated in Figure Continuing with the same example, now suppose that the source breaks the message into 5,000 packets, with each packet being 1,500 bits long. Again assuming that there is no congestion in the network, how long does it take to move the 5,000 packets from source to destination? As shown in Figure 1.11, it takes the source 1 msec to move the first packet from the source to the first switch. And it takes the first switch 1 msec to move this first packet from the first to the second switch. But while the first packet is being moved from the first switch to the second switch, the second packet is simultaneously moved from the source to the first switch. Thus the second packet reaches the first switch at time = 2 msec. Following this logic we see that the last packet is completely received at the first switch at time = 5,000 msec = 5 seconds. Since this last packet has to be transmitted on two more links, the last packet is received by the destination at seconds. Amazingly enough, message segmentation has reduced the end-to-end delay by a factor of three! But why is this so? What is packet switching doing that is different from message switching? The key difference is that message switching is performing sequential transmission whereas packet switching is performing parallel transmission (that is, pipelining). Observe that with message switching, while one node (the source or one of the switches) is transmitting, the remaining nodes are idle. With packet switching, once the first packet reaches the last switch, three nodes transmit at the same time.

24 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet Source Destination Packet switch Packet switch Time (msec) Key: packets Figure 1.11 Timing of message transfer when the message is segmented into 5,000 packets Message segmentation has yet another important advantage. As we will discuss later in this book, bit errors can be introduced into packets as they transit the network. When a switch detects an error in a packet, it typically discards the entire packet. So, if the entire message is transported as a single packet, and one bit in the message gets corrupted, the entire message is discarded. If, on the other hand, the message is segmented into many packets and one bit in one of the packets is corrupted, then only that one packet is discarded. Message segmentation does have some drawbacks, however. We will see that each packet must carry, in addition to the data being sent from the sending application to the receiving application, some control information. This information, which is carried in the packet header, might include the identity of the sender and receiver and a packet identifier (for example, a number). Since the amount of header information would be approximately the same for a message or a packet, the amount of header overhead per byte of data is higher for packet switching (with message segmentation) than for message switching. Before moving on to the next subsection, you are highly encouraged to explore the Message-Segmentation Java Applet that is available at the Web site for this book ( This applet will allow you to experiment with

25 C01 pp4 6/14/02 5:45 PM Page The Network Core 25 different message and packet sizes, and will allow you to examine the effect of additional propagation delays Packet Forwarding in Computer Networks There are two broad classes of packet-switched networks: datagram networks and virtual circuit networks. They differ in whether their switches use destination addresses or so-called virtual circuit numbers to forward packets toward their destinations. We ll call any network that forwards packets according to host destination addresses a datagram network. The routers in the Internet forward packets according to host destination addresses; hence the Internet is a datagram network. We ll call any network that forwards packets according to virtual circuit numbers a virtual circuit network. Examples of packet-switching technologies that use virtual circuits include X.25, frame relay, and ATM (asynchronous transfer mode). While the difference between using destination addresses and virtual circuit numbers may seem minor, the choice has a huge impact on how routes are set up and how routing is managed, as we ll see below. Virtual Circuit Networks A virtual circuit (VC) consists of (1) a path (that is, a series of links and packet switches) between the source and destination hosts, (2) virtual circuit numbers, one number for each link along the path, and (3) entries in VC-number translation tables in each packet switch along the path. Once a VC is established between source and destination, the source can send packets into the VC with the appropriate VC number. Because a VC has a different VC number on each link, an intermediate packet switch must replace the VC number of each traversing packet with a new one. The new VC number is obtained from the VC-number translation table. To illustrate the concept, consider the network shown in Figure Suppose host A requests that the network establish a VC between itself and host B. Suppose that the network chooses the path A-PS1-PS2-B and assigns VC numbers 12, 22, 32 to the three links in this path. Then, when a packet as in VC leaves host A, the A PS1 1 2 PS2 1 2 B 3 3 PS3 PS4 Figure 1.12 A simple virtual circuit network

26 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet value in the VC-number field is 12; when it leaves PS1, the value is 22; and when it leaves PS2, the value is 32. The numbers next to the links of PS1 are the interface numbers. How does the switch determine the replacement VC number for a packet traversing the switch? Each switch has a VC-number translation table; for example, the VC-number translation table in PS1 might look something like this: Incoming Interface Incoming VC # Outgoing Interface Outgoing VC # Whenever a new VC is established across a switch, an entry is added to the VCnumber table. Similarly, whenever a VC terminates, the entries in each table along its path are removed. You might be wondering why a packet doesn t just keep the same VC number on each of the links along its route. The answer is twofold. First, by replacing the number from link to link, the length of the VC field is reduced. Second, and more important, by permitting a different VC number for each link along the path of the VC, a network management function is simplified. Specifically, with multiple VC numbers, each link in the path can choose a VC number independently of what the other links in the path choose. If a common number were required for all links along the path, the switches would have to exchange and process a substantial number of messages to agree on the VC number to be used for a connection. If a network employs virtual circuits, then the network s switches must maintain state information for the ongoing connections. Specifically, each time a new connection is established across a switch, a new connection entry must be added to the switch s VC-number translation table; and each time a connection is released, an entry must be removed from the table. Note that even if there is no VC-number translation, it is still necessary to maintain state information that associates VC numbers to output interface numbers. The issue of whether or not a packet switch maintains state information for each ongoing connection is a crucial one one that we return to shortly below. Datagram Networks Datagram networks are analogous in many respects to the postal service. When a sender sends a letter to a destination, the sender wraps the letter in an envelope and writes the destination address on the envelope. This destination address has a hierarchical structure. For example, letters sent to a location in the United States include

27 C01 pp4 6/14/02 5:45 PM Page The Network Core 27 the country (USA), the state (for example, Pennsylvania), the city (for example, Philadelphia), the street (for example, Walnut Street), and the number of the house on the street (for example, 421). The postal service uses the address on the envelope to route the letter to its destination. For example, if the letter is sent from France, then a postal office in France will first forward the letter to a postal center in the United States. This postal center in the United States will then forward the letter to a postal center in Philadelphia. Finally, a mail carrier working in Philadelphia will deliver the letter to its ultimate destination. In a datagram network, each packet traversing the network contains in its header the address of the packet s destination. As with postal addresses, this address has a hierarchical structure. When a packet arrives at a packet switch in the network, the packet switch examines a portion of the packet s destination address and forwards the packet to an adjacent switch. More specifically, each packet switch has a forwarding table that maps destination addresses (or portions of the destination addresses) to an outbound link. When a packet arrives at a switch, the switch examines the address and indexes its table with this destination address to find the appropriate outbound link. The switch then directs the packet to this outbound link. The end-to-end routing process is also analogous to a car driver who does not use maps but instead prefers to ask for directions. For example, suppose Joe is driving from Philadelphia to 156 Lakeside Drive in Orlando, Florida. Joe first drives to his neighborhood gas station and asks how to get to 156 Lakeside Drive in Orlando, Florida. The gas station attendant extracts the Florida portion of the address and tells Joe that he needs to get onto the interstate highway I-95 South, which has an entrance just next to the gas station. He also tells Joe that once he enters Florida he should ask someone else there. Joe then takes I-95 South until he gets to Jacksonville, Florida, at which point he asks another gas station attendant for directions. The attendant extracts the Orlando portion of the address and tells Joe that he should continue on I-95 to Daytona Beach and then ask someone else. In Daytona Beach another gas station attendant also extracts the Orlando portion of the address and tells Joe that he should take I-4 directly to Orlando. Joe takes I-4 and gets off at the Orlando exit. Joe goes to another gas station attendant, and this time the attendant extracts the Lakeside Drive portion of the address and tells Joe the road he must follow to get to Lakeside Drive. Once Joe reaches Lakeside Drive, he asks a kid on a bicycle how to get to his destination. The kid extracts the 156 portion of the address and points to the house. Joe finally reaches his ultimate destination. We will be discussing packet forwarding in datagram networks in great detail in this book. But for now we mention that, in contrast with VC networks, datagram networks do not maintain connection-state information in their switches. In fact, a packet switch in a pure datagram network is completely oblivious to any flows of traffic that may be passing through it. A packet switch in a datagram network makes forwarding decisions based upon a packet s destination address, not upon the connection to which the packet belongs. Because VC networks must maintain connectionstate information in their switches, opponents of VC networks argue that VC networks

28 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet are overly complex. These opponents include most researchers and engineers in the Internet community. Proponents of VC networks feel that VCs can offer applications a richer variety of networking services. How would you actually like to see the end-to-end route that packets take in the Internet? We now invite you to get your hands dirty by interacting with the Traceroute program, by visiting the site (See a discussion of Traceroute in Section 1.6.) Network Taxonomy We have now introduced several important networking concepts: circuit switching, packet switching, virtual circuits, connectionless service, and connection-oriented service. How does it all fit together? First, in our simple view of the world, a telecommunication network either employs circuit switching or packet switching (see Figure 1.13). A link in a circuitswitched network can employ either FDM or TDM. Packet-switched networks are either virtual circuit networks or datagram networks. Switches in virtual circuit networks forward packets according to the packets VC numbers and maintain connection state. Switches in datagram networks forward packets according to the packets destination addresses and do not maintain connection state. A datagram network is not, however, either a connectionless or a connectionoriented network. Indeed, a datagram network can provide connectionless service to some of its applications and connection-oriented service to other applications. For example, the Internet, which is a datagram network, provides both connectionless and connection-oriented service to its applications. We saw in Section 1.2 that these services are provided in the Internet by the UDP and TCP protocols, respectively. Networks with VCs are, however, always connection-oriented. Telecommunications networks Circuit-switched networks Packet-switched networks FDM TDM Networks with VCs Datagram networks Figure 1.13 Taxonomy of telecommunication networks

29 C01 pp4 6/14/02 5:45 PM Page Network Access and Physical Media Network Access and Physical Media In Sections 1.2 and 1.3 we examined the roles of end systems and routers in a computer network. In this section we consider network access the physical link(s) that connect an end system to its edge router, which is the first router on a path from the end system to any other distant end system. Figure 1.14 shows several types of access links from end system to edge router; the access links are highlighted in thick, shaded lines. Since network access technology is closely tied to physical media technology (fiber, coaxial pair, twisted-pair telephone wire, radio spectrum), we consider these two topics together in this section. Local ISP Regional ISP Company Network Figure 1.14 Access networks

30 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet Network Access Network access can be loosely classified into three categories: Residential access, connecting home end systems into the network Company access, connecting end systems in a business or educational institution into the network Mobile access, connecting mobile end systems into the network These categories are not hard and fast for example, some company end systems may use the access technology that we ascribe to residential access, and vice versa. The following descriptions are meant to hold for the common cases. Residential Access Residential access refers to connecting a home end system (typically a PC, but perhaps a Web TV or some Internet home appliance) to an edge router. A prevalent form of residential access remains the dial-up modem over an ordinary analog telephone line into a residential ISP (such as America Online). The home modem converts the digital output of the PC into analog format for transmission over the analog phone line. This analog phone line is made of twisted-pair copper wire, and is the same telephone line used to make ordinary phone calls. (We will discuss twisted pair later in this section.) At the other end of the analog phone line, a modem in the ISP converts the analog signal back into digital form for input to the ISP router. Thus, the access network is simply a pair of modems along with a point-to-point dial-up phone line. Today s modem speeds allow dial-up access at rates up to 56 Kbps. However, due to the poor quality of the twisted-pair line between many homes and ISPs, many users get an effective rate significantly less than 56 Kbps. Many residential users find a dial-up modem s 56 Kbps access to be excruciatingly slow. For example, it takes approximately eight minutes to download a single three-minute MP3 song over a 56 Kbps dial-up modem. Moreover, dial-up modem access ties up a user s ordinary phone line while a residential user uses a dial-up modem to surf the Web, the user cannot receive and make ordinary phone calls over the phone line. Fortunately, new broadband access technologies are providing residential users higher bit rates; and they are also providing a means for users to access the Internet and talk on the phone at the same time. There are two common types of broadband residential access: digital subscriber line (DSL) [DSL 2002] and hybrid fiber coaxial cable (HFC) [Cable 2002]. As of January 2001, broadband residential access was much less prevalent than 56 Kbps dial-up modem access: the percentage of homes with broadband Internet access was 9.2 percent in South Korea, 4.2 percent in Canada, and 2.2 percent in the United States, with even lower penetration rates in Europe [Economist 2001]. However, DSL and HFC are being rapidly deployed throughout the world, with HFC be-

31 C01 pp4 6/14/02 5:45 PM Page Network Access and Physical Media 31 Coaxial cable Fiber cable Fiber node Hundreds of homes Head end Fiber node Hundreds of homes Figure 1.15 A hybrid fiber-coax access network ing generally more prevalent in the United States and DSL being generally more prevalent in Europe and Asia. DSL access is typically provided by a telephone company (for example, Pacific Bell or France Telecom), sometimes in partnership with an independent ISP. Conceptually similar to dial-up modems, DSL is a new modem technology again running over existing twisted-pair telephone lines. But by restricting the distance between user and ISP modem, DSL can transmit and receive data at much higher rates. The data rates are typically asymmetrical in the two directions, with a higher rate from ISP router to home than from the home to ISP router. The asymmetry in the data rates reflects the belief that a home user is more likely to be a consumer of information (bringing data into the home) than a producer of information. In theory, DSL can provide rates of more than 10 Mbps from ISP to home and more than 1 Mbps from home to ISP. However, in practice the rates offered by DSL providers are much less. As of 2002, typical downstream rates are 384 Kbps to 1.5 Mbps; and typical upstream rates are 128 Kbps to 256 Kbps. DSL uses frequency division multiplexing, as described in the previous section. In particular, DSL divides the communication link between the home and the ISP into three nonoverlapping frequency bands: A high-speed downstream channel, in the 50kHz to 1MHz band A medium-speed upstream channel, in the 4kHz to 50kHz band An ordinary two-way telephone channel, in the 0 to 4kHz band

32 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet The actual amount of downstream and upstream bandwidth potentially available to the user is a function of the distance between the home modem and the ISP modem, the gauge of the twisted-pair line, and the degree of electrical interference. In fact, unlike dial-up modems, engineers have explicitly designed DSL for short distances between residential and ISP modems, allowing for substantially higher transmission rates than dial-up access. While DSL and dial-up modems use ordinary phone lines, HFC access networks are extensions of the current cable network used for broadcasting cable television. In a traditional cable system, a cable head end broadcasts through a distribution network of coaxial cable and amplifiers to residences. As illustrated in Figure 1.15, fiber optics connect the cable head end to neighborhood-level junctions, from which traditional coaxial cable is then used to reach individual houses and apartments. Each neighborhood junction typically supports 500 to 5,000 homes. As with DSL, HFC requires special modems, called cable modems. Companies that provide cable Internet access require their customers to either purchase or lease a modem. Typically, the cable modem is an external device and connects to the home PC through a 10-BaseT Ethernet port. (We will discuss Ethernet in great detail in Chapter 5.) Cable modems divide the HFC network into two channels, a downstream and an upstream channel. As with DSL, the downstream channel is typically allocated more bandwidth and hence a faster transmission rate. However, with HFC (and not with DSL), these rates are shared among the homes, as we discuss next. One important characteristic of HFC is that it is a shared broadcast medium. In particular, every packet sent by the head end travels downstream on every link to every home; and every packet sent by a home travels on the upstream channel to the head end. For this reason, if several users are simultaneously downloading different MP3s on the downstream channel, the actual rate at which each user receives its MP3 will be significantly less than the downstream rate. On the other hand, if there are only a few active users and they are all Web surfing, then each of the users may actually receive Web pages at the full downstream rate, as users will rarely request a Web page at exactly the same time. Because the upstream channel is also shared, packets sent by two different homes at the same time will collide, which further decreases the effective upstream bandwidth. (We will discuss this collision issue in some detail when we discuss Ethernet in Chapter 5.) Advocates of DSL are quick to point out that DSL is a point-to-point connection between the home and ISP, and therefore all the DSL bandwidth is dedicated rather than shared. Cable advocates, however, argue that a reasonably dimensioned HFC network provides higher bandwidths than DSL. The battle between DSL and HFC for high-speed residential access has clearly begun. One of the attractive features of DSL and HFC is that the services are always on; that is, the user can leave his or her computer on and remain permanently connected to an ISP while simultaneously making and receiving ordinary telephone calls.

33 C01 pp4 6/14/02 5:45 PM Page Network Access and Physical Media 33 Company Access On corporate and university campuses, a local area network (LAN) is typically used to connect an end system to the edge router. As we will see in Chapter 5, there are many different types of LAN technology. However, Ethernet technology is currently by far the most prevalent access technology in company networks. Ethernet operates at 10 Mbps or 100 Mbps (and now even at 1 Gbps and 10 Gbps). It uses either twisted-pair copper wire or coaxial cable to connect a number of end systems with each other and with an edge router. The edge router is responsible for routing packets that have destinations outside of that LAN. Like HFC, Ethernet uses a shared medium, so that end users share the transmission rate of the LAN. More recently, shared Ethernet technology has been migrating toward switched Ethernet technology. Switched Ethernet uses multiple twisted-pair Ethernet segments connected at a switch to allow the full bandwidth of an Ethernet to be delivered to different users on the same LAN simultaneously. We will explore shared and switched Ethernet in detail in Chapter 5. Mobile Access Accompanying the current Internet revolution, the wireless revolution is also having a profound impact on the way people work and live. In the year 2000, more people in Europe had a mobile phone than had a PC or a car. And the wireless trend is continuing, with many analysts predicting that mobile handheld devices such as mobile phones and PDAs will overtake wired computers as the dominant Internet access devices throughout the world by 2004 [Dornan 2001]. Today, there are two broad types of wireless Internet access. In a wireless LAN, mobile users transmit/receive packets to/from a base station (also known as a wireless access point) within a radius of a few tens of meters. The base station is typically connected to the wired Internet and thus serves to connect wireless users to the wired network. In wide-area wireless access networks, the base station is managed by a telecommunications provider and serves users within a radius of tens of kilometers. Wireless LANs, based on IEEE b technology (also known as wireless Ethernet and Wi-Fi), are currently enjoying rapid deployment in university departments, business offices, coffee cafes, and homes. For example, Polytechnic University has installed IEEE b base stations on its Brooklyn campus, and all of Polytechnic s students are required to purchase a portable computer equipped with b. Using this wireless LAN infrastructure, students send and receive or surf the Web from anywhere on campus (for example, library, dorm room, classroom, or outdoor campus bench). The b technology, which we will discuss in detail in Chapter 5, provides a shared bandwidth of 11 Mbps. Today many homes are combining broadband residential access (that is, cable modems or DSL) with inexpensive wireless LAN technology to create powerful home networks. Figure 1.16 shows a schematic of a typical home network (actually,

34 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet House Cable headend Internet Figure 1.16 A schematic of a typical home network this is precisely the home network setup of one of the authors). This home network consists of a roaming laptop as well as a stationary PC; a base station (the wireless access point) which communicates with the roaming PC; a cable modem, providing the broadband access to the Internet; and a router, which interconnects the base station and the stationary PC with the cable modem. This network allows two household members to have broadband access to the Internet, with one member roaming from the kitchen to the backyard to the bedrooms. The total fixed cost for such a network is less than $500 (including the cable/dsl modem) and is dropping rapidly [Bricklin 2001]. When you access the Internet through wireless LAN technology, you typically need to be within a few tens of meters of a base station. This is feasible for home access, coffee shop access, and, more generally, access within and around a building. But what if you are on the beach or in your car and you need Internet access? For such wide-area access, roaming Internet users make use of the portable phone infrastructure, accessing base stations that are up to tens of kilometers away. WAP (wireless access protocol), widely available in Europe, and i-mode, widely available in Japan, are two technologies that allow for Internet access over the portable phone infrastructure. Resembling ordinary wireless phones but with somewhat bigger screens, WAP phones provide low-speed Internet access as well as portable phone service. Instead of HTML, WAP phones use a special markup language WML (WAP Markup Language) that has been optimized for small screens and low-speed access. In Europe, the WAP protocol runs on top of Europe s highly successful GSM wireless telephony infrastructure, which uses time-division multiplexing. WAP has

35 C01 pp4 6/14/02 5:45 PM Page Network Access and Physical Media 35 been a flop in Europe to date but is expected to become more popular when the new GPRS (General Packet Radio Service) packet technology is widely available in On the other hand, the proprietary i-mode technology, which is similar in concept and functionality to WAP, has been a huge success in Japan. Telecommunications companies are currently making enormous investments in 3G, standing for Third Generation wireless, which will provide packet-switched wide-area wireless Internet access at speeds in excess of 384 Kbps [Dornan 2001]. 3G systems should provide high-speed access to the Web and interactive video, and should provide voice quality that is better than that of an ordinary wired telephone. The first 3G systems have been deployed in Japan. With such huge investments being made in 3G technology, infrastructure, and licenses, many analysts (and investors!) wonder whether 3G will be the great success that it is hyped to be. Will it instead lose out to competing technologies such as IEEE ? The jury is still out. (See [Weinstein 2002] and the case history in Section 5.7.) Physical Media In the previous subsection, we gave an overview of some of the most important network access technologies in the Internet. As we described these technologies, we also indicated the physical media used. For example, we said that HFC uses a combination of fiber cable and coaxial cable. We said that dial-up 56 Kbps modems and ADSL use twisted-pair copper wire. And we said that mobile access networks use the radio spectrum. In this subsection we provide a brief overview of these and other transmission media that are commonly employed in the Internet. In order to define what is meant by a physical medium, let us reflect on the brief life of a bit. Consider a bit traveling from one end system, through a series of links and routers, to another end system. This poor bit gets transmitted many, many times! The source end system first transmits the bit, and shortly thereafter the first router in the series receives the bit; the first router then transmits the bit, and shortly thereafter the second router receives the bit; and so on. Thus our bit, when traveling from source to destination, passes through a series of transmitter-receiver pairs. For each transmitter-receiver pair, the bit is sent by propagating electromagnetic waves or optical pulses across a physical medium. The physical medium can take many shapes and forms and does not have to be of the same type for each transmitter-receiver pair along the path. Examples of physical media include twisted-pair copper wire, coaxial cable, multi-mode fiber-optic cable, terrestrial radio spectrum, and satellite radio spectrum. Physical media fall into two categories: guided media and unguided media. With guided media, the waves are guided along a solid medium, such as a fiber-optic cable, a twisted-pair copper wire, or a coaxial cable. With unguided media, the waves propagate in the atmosphere and in outer space, such as in a wireless LAN or a digital satellite channel. But before we get into the characteristics of the various media types, let us say a few words about their costs. The actual cost of the physical link (copper wire,

36 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet fiber-optic cable, and so on) is often relatively minor compared with other networking costs. In particular, the labor cost associated with the installation of the physical link can be orders of magnitude higher than the cost of the material. For this reason, many builders install twisted pair, optical fiber, and coaxial cable in every room in a building. Even if only one medium is initially used, there is a good chance that another medium could be used in the near future, and so money is saved by not having to lay additional wires in the future. Twisted-Pair Copper Wire The least expensive and most commonly used guided transmission medium is twisted-pair copper wire. For over 100 years it has been used by telephone networks. In fact, more than 99 percent of the wired connections from the telephone handset to the local telephone switch use twisted-pair copper wire. Most of us have seen twisted pair in our homes and work environments. Twisted pair consists of two insulated copper wires, each about 1 mm thick, arranged in a regular spiral pattern. The wires are twisted together to reduce the electrical interference from similar pairs close by. Typically, a number of pairs are bundled together in a cable by wrapping the pairs in a protective shield. A wire pair constitutes a single communication link. Unshielded twisted pair (UTP) is commonly used for computer networks within a building, that is, for local area networks (LANs). Data rates for LANs using twisted pair today range from 10 Mbps to 1 Gbps. The data rates that can be achieved depend on the thickness of the wire and the distance between transmitter and receiver. Two types of UTP are common in LANs: category 3 and category 5. Category 3 corresponds to voice-grade twisted pair, commonly found in office buildings. Office buildings are often prewired with two or more parallel pairs of category 3 twisted pair; one pair is used for telephone communication, and the additional pairs can be used for additional telephone lines or for LAN networking. 10 Mbps Ethernet, one of the most prevalent LAN types, can use category 3 UTP. Category 5, with its more twists per centimeter and Teflon insulation, can handle higher bit rates. 100 Mbps Ethernet running on category 5 UTP has become very popular in recent years. Category 5 UTP has become common for preinstallation in new office buildings. When fiber-optic technology emerged in the 1980s, many people disparaged twisted pair because of its relatively low bit rates. Some people even felt that fiberoptic technology would completely replace twisted pair. But twisted pair did not give up so easily. Modern twisted-pair technology, such as category 5 UTP, can achieve data rates of 100 Mbps for distances up to a few hundred meters. Even higher rates are possible over shorter distances. In the end, twisted pair has emerged as the dominant solution for high-speed LAN networking. As discussed in the section on access networks, twisted pair is also commonly used for residential Internet access. We saw that dial-up modem technology enables access at rates of up to 56 Kbps over twisted pair. We also saw that DSL (digital sub-

37 C01 pp4 6/14/02 5:45 PM Page Network Access and Physical Media 37 scriber line) technology has enabled residential users to access the Internet at rates in excess of 6 Mbps over twisted pair (when users live close to the ISP s modem). Coaxial Cable Like twisted pair, coaxial cable consists of two copper conductors, but the two conductors are concentric rather than parallel. With this construction and a special insulation and shielding, coaxial cable can have high bit rates. Coaxial cable comes in two varieties: baseband coaxial cable and broadband coaxial cable. Baseband coaxial cable, also called 50-ohm cable, is about a centimeter thick, lightweight, and easy to bend. It is commonly used in LANs; in fact, the computer you use at work or at school may be connected to a LAN with either baseband coaxial cable or with UTP. Take a look at the connection to your computer s interface card. If you see a telephone-like jack and some wire that resembles telephone wire, you are using UTP; if you see a T-connector and a cable running out of both sides of the T-connector, you are using baseband coaxial cable. The term baseband comes from the fact that the stream of bits is dumped directly into the cable, without shifting the signal to a different frequency band. 10 Mbps Ethernets can use either UTP or baseband coaxial cable. However, almost all new Ethernet installments use UTP, rendering Ethernet with coaxial cable a legacy technology. Broadband coaxial cable, also called 75-ohm cable, is quite a bit thicker, heavier, and stiffer than the baseband variety. Broadband cable is quite common in cable television systems. As we saw earlier, cable television systems have recently been coupled with cable modems to provide residential users with Internet access at rates of 1 Mbps or higher. With broadband coaxial cable, the transmitter shifts the digital signal to a specific frequency band, and the resulting analog signal is sent from the transmitter to one or more receivers. Both baseband and broadband coaxial cable can be used as a guided shared medium. Specifically, a number of end systems can be connected directly to the cable, and all the end systems receive whatever is sent by the other end systems. Fiber Optics An optical fiber is a thin, flexible medium that conducts pulses of light, with each pulse representing a bit. A single optical fiber can support tremendous bit rates, up to tens or even hundreds of gigabits per second. They are immune to electromagnetic interference, have very low signal attenuation up to 100 kilometers, and are very hard to tap. These characteristics have made fiber optics the preferred longhaul guided transmission media, particularly for overseas links. Many of the longdistance telephone networks in the United States and elsewhere now use fiber optics exclusively. Fiber optics is also prevalent in the backbone of the Internet. However, the high cost of optical devices such as transmitters, receivers, and switches has hindered their deployment for short-haul transport, such as in a LAN or into the

38 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet home in a residential access network. [Goralski 2001], [Ramaswami 1998], and [Green 1992] provide in-depth coverage of optical networks. Terrestrial Radio Channels Radio channels carry signals in the electromagnetic spectrum. They are an attractive medium because they require no physical wire to be installed, can penetrate walls, provide connectivity to a mobile user, and can potentially carry a signal for long distances. The characteristics of a radio channel depend significantly on the propagation environment and the distance over which a signal is to be carried. Environmental considerations determine path loss and shadow fading (which decrease the signal strength as the signal travels over a distance and around/through obstructing objects), multipath fading (due to signal reflection off of interfering objects), and interference (due to other radio channels or electromagnetic signals). Terrestrial radio channels can be broadly classified into two groups: those that operate in local areas, typically spanning from ten to a few hundred meters; and those that operate in the wide area, spanning tens of kilometers. The wireless LAN products described in Section use local-area radio channels; WAP, i-mode, and 3G technologies, also discussed in Section 1.4.1, use wide-area radio channels. See [Dornan 2001] for a survey and discussion of the technology and products. Satellite Radio Channels A communication satellite links two or more Earth-based microwave transmitter/receivers, known as ground stations. The satellite receives transmissions on one frequency band, regenerates the signal using a repeater (discussed below), and transmits the signal on another frequency. Satellites can provide bandwidths in the gigabit per second range. Two types of satellites are used in communications: geostationary satellites and low-altitude satellites. Geostationary satellites permanently remain above the same spot on Earth. This stationary presence is achieved by placing the satellite in orbit at 36,000 kilometers above Earth s surface. This huge distance from ground station through satellite back to ground station introduces a substantial signal propagation delay of 250 milliseconds. Nevertheless, satellite links, which can operate at speeds of hundreds of Mbps, are often used in telephone networks and in the backbone of the Internet. Low-altitude satellites are placed much closer to Earth and do not remain permanently above one spot on Earth. They rotate around Earth just as the Moon does. To provide continuous coverage to an area, many satellites need to be placed in orbit. There are currently many low-altitude communication systems in development. Lloyd s satellite constellation Web page [Wood 2002] provides and collects information on satellite constellation systems for communications. Low-altitude satellite technology may be used for Internet access sometime in the future.

39 C01 pp4 6/14/02 5:45 PM Page ISPs amd Internet Backbones ISPs and Internet Backbones We saw earlier that end systems (user PCs, PDAs, Web servers, mail servers, and so on) connect into the Internet via an access network. Recall that the access network may be a wired or wireless local area network (for example, in a company, school, or library) or may be a residential Internet Service Provider (for example, AOL or MSN) that is reached via dial-up modem, cable modem, or DSL. But connecting end users and content providers into access networks is only a small piece of solving the puzzle of connecting together the hundreds of millions of users and hundreds of thousands of networks that make up the Internet. The Internet is a network of networks understanding this phrase will be the key to solving this puzzle. In the public Internet, access networks situated at the edge of the Internet are connected to the rest of the Internet through a tiered hierarchy of Internet Service Providers (ISPs), as shown in Figure Access ISPs (for example, residential ISPs such as AOL, and company ISPs using LANs) are at the bottom of this hierarchy. At the very top of the hierarchy is a relatively small number of so-called tier-1 ISPs. In many ways, a tier-1 ISP is the same as any network it has links and routers and is connected to other networks. In other ways, however, tier-1 ISPs are special. Their link speeds are often 622 Mbps or higher, with the larger tier-1 ISPs having links in the Gbps range; their routers must consequently be able to forward packets at extremely high rates. Tier-1 ISPs are also characterized by being: Directly connected to each of the other tier-1 ISPs Connected to a large number of tier-2 ISPs and other customer networks International in coverage Tier-1 ISPs are also known as Internet backbone networks. As of this writing, UUNet (a subsidiary of Worldcom) is the largest tier-1 ISP; other major backbone companies include Sprint, AT&T, Genuity, and Cable and Wireless. A tier-2 ISP typically has regional or national coverage, and (importantly) connects to only a few of the tier-1 ISPs (see Figure 1.17). Thus, in order to reach a large portion of the global Internet, a tier-2 ISP needs to route traffic through one of the tier-1 ISPs to which it is connected. A tier-2 ISP is said to be a customer of the tier-1 ISPs to which it is connected, and the tier-1 ISP is said to be a provider to its customer. Many large companies and institutions connect their enterprise s network directly into a tier-1 or tier-2 ISP, thus becoming a customer of that ISP. A provider ISP charges its customer ISP a fee, which typically depends on the bandwidth of the link connecting the two. A tier-2 network may also choose to connect directly to other tier-2 networks, in which case traffic can flow between the two tier-2 networks without having to pass through a tier-1 network. Below the tier-2 ISPs are the lowertier ISPs, which connect to the larger Internet via one or more tier-2 ISPs. At the bottom of the hierarchy are the access ISPs. Further complicating matters, some tier-1

40 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet Tier 3 Tier 2 Tier 1 Figure 1.17 Interconnection of ISPs providers are also tier-2 providers (that is, vertically integrated), selling Internet access directly to end users and content providers, as well as to lower-tier ISPs. When two ISPs are directly connected to each other, they are said to peer with each other. Within an ISP s network, the points at which the ISP connect to other ISPs (whether below, above, or at the same level in the hierarchy) are known as Points of Presence (POPs). A POP is simply a group of one or more routers in the ISP s network at which routers in other ISPs can connect. A tier-1 provider typically has many POPs scattered across different geographical locations in its network, with multiple customer ISPs connecting into each POP. For a customer ISP to connect to a provider s POP, the customer ISP typically leases a high-speed link from a third-party telecommunications provider and directly connects one of its routers to a router at the provider s POP. Two tier-1 ISPs can also peer with each other by connecting together a pair of POPs, one from each of the two ISPs. Furthermore, two ISPs may have multiple peering points connecting with each other at two or more pairs of POPs. In addition to connecting to each other at private peering points, ISPs often interconnect at Network Access Points (NAPs), each of which can be owned and op-

41 C01 pp4 6/14/02 5:45 PM Page Delay and Loss in Packet-Switched Networks 41 erated by either some third-party telecommunications company or by an Internet backbone provider. NAPs exchange huge quantities of traffic among many ISPs. However, increasingly, tier-1 ISPs are bypassing the NAPs and are interconnecting directly at private peering points [Kende 2000]. The trend is for the tier-1 ISPs to interconnect with each other directly at private peering points, and for tier-2 ISPs to interconnect with other tier-2 ISPs and with tier-1 ISPs at NAPs. Because the NAPs relay and switch tremendous volumes of traffic, they are in themselves complex high-speed switching networks, often concentrated in a single building. Often a NAP uses high-speed ATM switching technology in the heart of the NAP, with IP riding on top of ATM. (ATM is discussed in Chapter 5.) In summary, the topology of the Internet is complex, consisting of dozens of tier-1 and tier-2 ISPs and thousands of lower-tier ISPs. The ISPs are diverse in their coverage, with some spanning multiple continents and oceans, and others limited to narrow regions of the world. The lower-tier ISPs connect to the higher-tier ISPs, and the higher-tier ISPs interconnect at (typically) private peering points and NAPs. Users and content providers are customers of lower-tier ISPs, and lower-tier ISPs are customers of higher-tier ISPs. We conclude this section by mentioning that any one of us can become an access ISP as soon as we have an Internet connection. All we need to do is purchase the necessary equipment (for example, router and modem pool) to allow other users to connect to us. Thus new tiers and branches can be added to the Internet topology just as a new piece of Lego can be attached to an existing Lego construction. 1.6 Delay and Loss in Packet-Switched Networks Having now briefly considered the major pieces of the Internet architecture the applications, end systems, end-to-end transport protocols, routers, and links let us now consider what can happen to a packet as it travels from its source to its destination. Recall that a packet starts in a host (the source), passes through a series of routers, and ends its journey in another host (the destination). As a packet travels from one node (host or router) to the subsequent node (host or router) along this path, the packet suffers from several different types of delays at each node along the path. The most important of these delays are the nodal processing delay, queuing delay, transmission delay, and propagation delay; together, these delays accumulate to give a total nodal delay. In order to acquire a deep understanding of packet switching and computer networks, we must understand the nature and importance of these delays Types of Delay Let us explore these delays in the context of Figure As part of its end-to-end route between source and destination, a packet is sent from the upstream node

42 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet A B Nodal processing Propagation Queueing Transmission (waiting for transmission) Figure 1.18 The nodal delay at router A through router A to router B. Our goal is to characterize the nodal delay at router A. Note that router A has an outbound link leading to router B. This link is preceded by a queue (also known as a buffer). When the packet arrives at router A from the upstream node, router A examines the packet s header to determine the appropriate outbound link for the packet, and then directs the packet to the link. In this example, the outbound link for the packet is the one that leads to router B. A packet can be transmitted on a link only if there is no other packet currently being transmitted on the link and if there are no other packets preceding it in the queue; if the link is currently busy or if there are other packets already queued for the link, the newly arriving packet will then join the queue. Processing Delay The time required to examine the packet s header and determine where to direct the packet is part of the processing delay. The processing delay can also include other factors, such as the time needed to check for bit-level errors in the packet that occurred in transmitting the packet s bits from the upstream node to router A. Processing delays in high-speed routers are typically on the order of microseconds or less. After this nodal processing, the router directs the packet to the queue that precedes the link to router B. (In Chapter 4 we will study the details of how a router operates.) Queuing Delay At the queue, the packet experiences a queuing delay as it waits to be transmitted onto the link. The queuing delay of a specific packet will depend on the number of earlier-arriving packets that are queued and waiting for transmission across the link. The delay of a given packet can vary significantly from packet to packet. If the queue is empty and no other packet is currently being transmitted, then our packet s queuing delay is zero. On the other hand, if the traffic is heavy and many other pack-

43 C01 pp4 6/14/02 5:45 PM Page Delay and Loss in Packet-Switched Networks 43 ets are also waiting to be transmitted, the queuing delay will be long. We will see shortly that the number of packets that an arriving packet might expect to find on arrival is a function of the intensity and nature of the traffic arriving at the queue. Queuing delays can be on the order of microseconds to milliseconds in practice. Transmission Delay Assuming that packets are transmitted in first-come-first-served manner, as is common in packet-switched networks, our packet can be transmitted only after all the packets that have arrived before it have been transmitted. Denote the length of the packet by L bits, and denote the transmission rate of the link from router A to router B by R bits/sec. The rate R is determined by the transmission rate of the link to router B. For example, for a 10 Mbps Ethernet link, the rate is R 5 10 Mbps; for a 100 Mbps Ethernet link, the rate is R Mbps. The transmission delay (also called the store-and-forward delay, as discussed in Section 1.3) is L/R. This is the amount of time required to push (that is, transmit) all of the packet s bits into the link. Transmission delays are typically on the order of microseconds to milliseconds in practice. Propagation Delay Once a bit is pushed onto the link, it needs to propagate to router B. The time required to propagate from the beginning of the link to router B is the propagation delay. The bit propagates at the propagation speed of the link. The propagation speed depends on the physical medium of the link (that is, fiber optics, twisted-pair copper wire, and so on) and is in the range of meters/sec to meters/sec which is equal to, or a little less than, the speed of light. The propagation delay is the distance between two routers divided by the propagation speed. That is, the propagation delay is d/s, where d is the distance between router A and router B and s is the propagation speed of the link. Once the last bit of the packet propagates to node B, it and all the preceding bits of the packet are stored in router B. The whole process then continues with router B now performing the forwarding. In wide-area networks, propagation delays are on the order of milliseconds. Comparing Transmission and Propagation Delay Newcomers to the field of computer networking sometimes have difficulty understanding the difference between transmission delay and propagation delay. The difference is subtle but important. The transmission delay is the amount of time required for the router to push out the packet; it is a function of the packet s length and the transmission rate of the link, but has nothing to do with the distance between

44 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet the two routers. The propagation delay, on the other hand, is the time it takes a bit to propagate from one router to the next; it is a function of the distance between the two routers, but has nothing to do with the packet s length or the transmission rate of the link. An analogy might clarify the notions of transmission and propagation delay. Consider a highway that has a toll booth every 100 kilometers. You can think of the highway segments between toll booths as links and the toll booths as routers. Suppose that cars travel (that is, propagate) on the highway at a rate of 100 km/hour (that is, when a car leaves a toll booth, it instantaneously accelerates to 100 km/hour and maintains that speed between toll booths). Suppose that there is a caravan of 10 cars that are traveling together, and that these 10 cars follow each other in a fixed order. You can think of each car as a bit and the caravan as a packet. Also suppose that each toll booth services (that is, transmits) a car at a rate of 1 car per 12 seconds, and that it is late at night so that the caravan s cars are the only cars on the highway. Finally, suppose that whenever the first car of the caravan arrives at a toll booth, it waits at the entrance until the 9 other cars have arrived and lined up behind it. (Thus the entire caravan must be stored at the toll booth before it can begin to be forwarded. ) The time required for the toll booth to push the entire caravan onto the highway is (10 cars)/(5 cars/minute) 5 2 minutes. This time is analogous to the transmission delay in a router. The time required for a car to travel from the exit of one toll booth to the next toll booth is 100 km/(100 km/hour) 5 1 hour. This time is analogous to propagation delay. Therefore, the time from when the caravan is stored in front of a toll booth until the caravan is stored in front of the next toll booth is the sum of transmission delay and propagation delay in this example, 62 minutes. Let s explore this analogy a bit more. What would happen if the toll-booth service time for a caravan were greater than the time for a car to travel between toll booths? For example, suppose now that cars travel instead at the rate of 1,000 km/hour and the toll booth services cars at the rate of one car per minute. Then the traveling delay between two toll booths is 6 minutes and the time to serve a caravan is 10 minutes. In this case, the first few cars in the caravan will arrive at the second toll booth before the last cars in the caravan leave the first toll booth. This situation also arises in packet-switched networks the first bits in a packet can arrive at a router while many of the remaining bits in the packet are still waiting to be transmitted by the preceding router. If we let d proc,d queue, d trans, and d prop denote the processing, queuing, transmission, and propagation delays, then the total nodal delay is given by d nodal 5 d proc + d queue + d trans + d prop The contribution of these delay components can vary significantly. For example, d prop can be negligible (for example, a couple of microseconds) for a link connecting two routers on the same university campus; however, d prop is hundreds of milliseconds for two routers interconnected by a geostationary satellite link, and can be the

45 C01 pp4 6/14/02 5:45 PM Page Delay and Loss in Packet-Switched Networks 45 dominant term in d nodal. Similarly, d trans can range from negligible to significant. Its contribution is typically negligible for transmission rates of 10 Mbps and higher (for example, for LANs); however, it can be hundreds of milliseconds for large Internet packets sent over low-speed dial-up modem links. The processing delay, d proc,is often negligible; however, it strongly influences a router s maximum throughput, which is the maximum rate at which a router can forward packets Queuing Delay and Packet Loss The most complicated and interesting component of nodal delay is the queuing delay, d queue. In fact, queuing delay is so important and interesting in computer networking that thousands of papers and numerous books have been written about it [Bertsekas 1991; Daigle 1991; Kleinrock 1975, 1976; Ross 1995]! We give only a high-level, intuitive discussion of queuing delay here; the more curious reader may want to browse through some of the books (or even eventually write a Ph.D. thesis on the subject!). Unlike the other three delays (namely, d proc, d trans,and d prop ), the queuing delay can vary from packet to packet. For example, if ten packets arrive at an empty queue at the same time, the first packet transmitted will suffer no queuing delay, while the last packet transmitted will suffer a relatively large queuing delay (while it waits for the other nine packets to be transmitted). Therefore, when characterizing queuing delay, one typically uses statistical measures, such as average queuing delay, variance of queuing delay, and the probability that the queuing delay exceeds some specified value. When is the queuing delay large and when is it insignificant? The answer to this question depends largely on the rate at which traffic arrives at the queue, the transmission rate of the link, and the nature of the arriving traffic, that is, whether the traffic arrives periodically or whether it arrives in bursts. To gain some insight here, let a denote the average rate at which packets arrive at the queue (a is in units of packets/sec). Recall that R is the transmission rate; that is, it is the rate (in bits/sec) at which bits are pushed out of the queue. Also suppose, for simplicity, that all packets consist of L bits. Then the average rate at which bits arrive at the queue is La bits/sec. Finally, assume that the queue is very big, so that it can hold essentially an infinite number of bits. The ratio La/R, called the traffic intensity, often plays an important role in estimating the extent of the queuing delay. If La/R. 1, then the average rate at which bits arrive at the queue exceeds the rate at which the bits can be transmitted from the queue. In this unfortunate situation, the queue will tend to increase without bound and the queuing delay will approach infinity! Therefore, one of the golden rules in traffic engineering is: Design your system so that the traffic intensity is no greater than 1. Now consider the case La/R # 1. Here, the nature of the arriving traffic impacts the queuing delay. For example, if packets arrive periodically that is, one packet arrives every L/R seconds then every packet will arrive at an empty queue and there will be no queuing delay. On the other hand, if packets arrive in bursts but periodically,

46 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet there can be a significant average queuing delay. For example, suppose N packets arrive simultaneously every (L/R)N seconds. Then the first packet transmitted has no queuing delay; the second packet transmitted has a queuing delay of L/R seconds; and more generally, the nth packet transmitted has a queuing delay of (n 1)L/R seconds. We leave it as an exercise for the reader to calculate the average queuing delay in this example. The two examples described above of periodic arrivals are a bit academic. Typically, the arrival process to a queue is random; that is, the arrivals do not follow any pattern and the packets are spaced apart by random amounts of time. In this more realistic case, the quantity La/R is not usually sufficient to fully characterize the delay statistics. Nonetheless, it is useful in gaining an intuitive understanding of the extent of the queuing delay. In particular, if traffic intensity is close to zero, then packet arrivals are few and far between and it is unlikely that an arriving packet will find another packet in the queue. Hence, the average queuing delay will be close to zero. On the other hand, when the traffic intensity is close to 1, there will be intervals of time when the arrival rate exceeds the transmission capacity (due to the burstiness of arrivals), and a queue will form. As the traffic intensity approaches 1, the average queue length gets larger and larger. The qualitative dependence of average queuing delay on the traffic intensity is shown in Figure One important aspect of Figure 1.19 is the fact that as the traffic intensity approaches 1, the average queuing delay increases rapidly. A small percentage increase in the intensity will result in a much larger percentage-wise increase in delay. Perhaps you have experienced this phenomenon on the highway. If you regularly drive on a road that is typically congested, the fact that the road is typically congested means that its traffic intensity is close to 1. If some event causes an even slightly larger-than-usual amount of traffic, the delays you experience can be huge. Average queueing delay La/R 1 Figure 1.19 Dependence of average queuing delay on traffic intensity

47 C01 pp4 6/14/02 5:45 PM Page Delay and Loss in Packet-Switched Networks 47 Packet Loss In our discussions above, we have assumed that the queue is capable of holding an infinite number of packets. In reality a queue preceding a link has finite capacity, although the queuing capacity greatly depends on the switch design and cost. Because the queue capacity is finite, packet delays do not really approach infinity as the traffic intensity approaches 1. Instead, a packet can arrive to find a full queue. With no place to store such a packet, a router will drop that packet; that is, the packet will be lost. From an end-system viewpoint, this will look like a packet having been transmitted into the network core but never emerging from the network at the destination. The fraction of lost packets increases as the traffic intensity increases. Therefore, performance at a node is often measured not only in terms of delay, but also in terms of the probability of packet loss. As we shall discuss in the subsequent chapters, a lost packet may be retransmitted on an end-to-end basis, either by the application or by the transport layer protocol. End-to-End Delay Our discussion up to this point has been focused on the nodal delay, that is, the delay at a single router. Let us conclude our discussion by briefly considering the delay from source to destination. To get a handle on this concept, suppose there are N 1 routers between the source host and the destination host. Let us also suppose that the network is uncongested (so that queuing delays are negligible), the processing delay at each router and at the source host is d proc, the transmission rate out of each router and out of the source host is R bits/sec, and the propagation on each link is d prop. The nodal delays accumulate and give an end-to-end delay, d end-end 5 N (d proc + d trans + d prop ) where, once again, d trans 5 L/R, where L is the packet size. We leave it to the reader to generalize this formula to the case of heterogeneous delays at the nodes and to the presence of an average queuing delay at each node Delay and Routes in the Internet To get a hands-on feel for the delay in a computer network, we can make use of the Traceroute diagnostic program. Traceroute is a simple program that can run in any Internet host. When the user specifies a destination host name, the program in the source host sends multiple, special packets toward that destination. As these packets work their way toward the destination, they pass through a series of routers. When a router receives one of these special packets, it sends a short message back to the source. This message contains the name and address of the router. More specifically, suppose there are N 1 routers between the source and the destination. Then the source will send N special packets into the network, with each

48 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet packet addressed to the ultimate destination. These N special packets are marked 1 through N, with the first packet marked 1 and the last packet marked N. When the nth router receives the nth packet marked n, the router destroys the packet and sends a message back to the source. And when the destination host receives the Nth packet, the destination destroys it as well, but again returns a message back to the source. The source records the time that elapses from when it sends a packet until when it receives the corresponding return message; it also records the name and address of the router (or the destination host) that returns the message. In this manner, the source can reconstruct the route taken by packets flowing from source to destination, and the source can determine the round-trip delays to all the intervening routers. Traceroute actually repeats the experiment just described three times, so the source actually sends 3 N packets to the destination. RFC 1393 describes Traceroute in detail. Here is an example of the output of the Traceroute program, where the route was being traced from the source host eniac.seas.upenn.edu (at the University of Pennsylvania) to diane.ibp.fr (at the University of Paris VI). The output has six columns: the first column is the n value described above, that is, the number of the router along the route; the second column is the name of the router; the third column is the address of the router (of the form xxx.xxx.xxx.xxx); the last three columns are the round-trip delays for three experiments. If the source receives fewer than three messages from any given router (due to packet loss in the network), Traceroute places an asterisk just after the router number and reports fewer than three roundtrip times for that router. 1 GW.CIS.UPENN.EDU ( ) 3 ms 2 ms 1 ms 2 DEFAULT7-GW.UPENN.EDU ( ) 3 ms 1 ms 2 ms ( ) 3 ms 4 ms 3 ms 4 border2-hssi1-0.westorange.mci.net ( ) 6 ms 6 ms 6 ms 5 core1-fddi-1.westorange.mci.net ( ) 7 ms 6 ms 6 ms 6 somerouter.sprintlink.net ( ) 16 ms 305 ms 192 ms 7 somerouter.sprintlink.net ( ) 20 ms 196 ms 18 ms 8 sl-dc-6-h2/0-t3.sprintlink.net ( ) 19 ms 18 ms 24 ms ( ) 19 ms 24 ms 18 ms 10 gsl-dc-3-fddi0/0.gsl.net ( ) 19 ms 18 ms 20 ms 11 * raspail-ip.eurogate.net ( ) 133 ms 94 ms 12 raspail-ip2.eurogate.net ( ) 93 ms 95 ms 97 ms

49 C01 pp4 6/14/02 5:45 PM Page Protocol Layers and Their Service Models ( ) 200 ms 94 ms 209 ms 14 stamand1.renater.ft.net ( ) 105 ms 101 ms 105 ms 15 stlambert.rerif.ft.net ( ) 108 ms 102 ms 95 ms 16 danton1.rerif.ft.net ( ) 110 ms 97 ms 91 ms 17 u-jussieu-paris.rerif.ft.net ( ) 94 ms 96 ms 100 ms 18 r-jusren.reseau.jussieu.fr ( ) 100 ms 94 ms 100 ms 19 r-ibp.reseau.jussieu.fr ( ) 96 ms 100 ms 94 ms 20 masi.ibp.fr ( ) 121 ms 100 ms 97 ms 21 * diane.ibp.fr ( ) 105 ms 102 ms In the trace above there are 20 routers between the source and the destination. Most of these routers have a name, and all of them have addresses. For example, the name of router 8 is sl-dc-6-h2/0-t3.sprintlink.net and its address is Looking at the data provided for this same router, we see that in the first of the three trials the round-trip delay between the source and the router 8 was 19 msec. The roundtrip delays for the subsequent two trials were 18 and 24 msec. These round-trip delays include all of the delays just discussed, including transmission delays, propagation delays, router processing delays, and queuing delays. Because the queuing delay is varying with time, the round-trip delay of packet n sent to a router n can actually be longer than the round-trip delay of packet n+1 sent to router n+1. Note in the example above that there is a big jump in the round-trip delay when going from router 10 to router 11. This is because the link between routers 10 and 11 is a transatlantic link. Want to try out Traceroute for yourself? It is highly recommended that you visit which provides a Web interface to an extensive of list of sources for route tracing. You choose a source and supply the hostname for any destination. The Traceroute program then does all the work. 1.7 Protocol Layers and Their Service Models From our discussion thus far, it is apparent that the Internet is an extremely complicated system. We have seen that there are many pieces to the Internet: numerous applications and protocols, various types of end systems and connections between end systems, routers, and various types of link-level media. Given this enormous complexity, is there any hope of organizing network architecture, or at least our discussion of network architecture? Fortunately, the answer to both questions is yes.

50 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet Layered Architecture Before attempting to organize our thoughts on Internet architecture, let s look for a human analogy. Actually, we deal with complex systems all the time in our everyday life. Imagine if someone asked you to describe, for example, the airline system. How would you find the structure to describe this complex system that has ticketing agents, baggage checkers, gate personnel, pilots, airplanes, air traffic control, and a worldwide system for routing airplanes? One way to describe this system might be to describe the series of actions you take (or others take for you) when you fly on an airline. You purchase your ticket, check your bags, go to the gate, and eventually get loaded onto the plane. The plane takes off and is routed to its destination. After your plane lands, you deplane at the gate and claim your bags. If the trip was bad, you complain about the flight to the ticket agent (getting nothing for your effort). This scenario is shown in Figure Already, we can see some analogies here with computer networking: You are being shipped from source to destination by the airline; a packet is shipped from source host to destination host in the Internet. But this is not quite the analogy we are after. We are looking for some structure in Figure Looking at Figure 1.20, we note that there is a ticketing function at each end; there is also a baggage function for already-ticketed passengers, and a gate function for already-ticketed and already-baggage-checked passengers. For passengers who have made it through the gate (that is, passengers who are already ticketed, baggage-checked, and through the gate), there is a takeoff and landing function, and while in flight, there is an airplanerouting function. This suggests that we can look at the functionality in Figure 1.20 in a horizontal manner, as shown in Figure Ticket (purchase) Baggage (check) Gates (load) Runway takeoff Airplane routing Ticket (complain) Baggage (claim) Gates (unload) Runway landing Airplane routing Airplane routing Figure 1.20 Taking an airplane trip: Actions

51 C01 pp4 6/14/02 5:45 PM Page Protocol Layers and Their Service Models 51 Ticket (purchase) Baggage (check) Gates (load) Runway takeoff Ticket (complain) Baggage (claim) Gates (unload) Runway landing Ticket Baggage Gate Takeoff/Landing Airplane routing Airplane routing Airplane routing Airplane routing Figure 1.21 Horizontal layering of airline functionality Figure 1.21 has divided the airline functionality into layers, providing a framework in which we can discuss airline travel. Now, when we want to describe a part of airline travel, we can talk about a specific, well-defined component of airline travel. For example, when we discuss gate functionality, we know we are discussing functionality that sits below baggage handling, and above takeoff and landing. We note that each layer, combined with the layers below it, implements some functionality, some service. At the ticketing layer and below, airline-counter-to-airlinecounter transfer of a person is accomplished. At the baggage layer and below, baggage-check-to-baggage-claim transfer of a person and bags is accomplished. Note that the baggage layer provides this service only to an already-ticketed person. At the gate layer, departure-gate-to-arrival-gate transfer of a person and bags is accomplished. At the takeoff/landing layer, runway-to-runway transfer of people and their bags is accomplished. Each layer provides its service by (1) performing certain actions within that layer (for example, at the gate layer, loading and unloading people from an airplane) and by (2) using the services of the layer directly below it (for example, in the gate layer, using the runway-to-runway passenger transfer service of the takeoff/landing layer). As noted above, a layered architecture allows us to discuss a well-defined, specific part of a large and complex system. This simplification itself is of considerable value. When a system has a layered structure it is also much easier to change the implementation of the service provided by the layer. As long as the layer provides the same service to the layer above it, and uses the same services from the layer below it, the remainder of the system remains unchanged when a layer s implementation is changed. (Note that changing the implementation of a service is very different from changing the service itself!) For example, if the gate functions were changed (for instance, to have people board and disembark by height), the remainder of the airline system would remain unchanged since the gate layer still provides the same function (loading and unloading people); it simply implements that function in a different manner after the change. For large and complex systems that are constantly

52 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet being updated, the ability to change the implementation of a service without affecting other components of the system is another important advantage of layering. But enough with airlines. Let s now turn our attention to network protocols. To reduce design complexity, network designers organize protocols and the network hardware and software that implement the protocols in layers. With a layered protocol architecture, each protocol belongs to one of the layers. It s important to realize that a protocol in layer n is distributed among the network entities (including end systems and packet switches) that implement that protocol, just as the functions in our layered airline architecture were distributed between the departing and arriving airports. In other words, there s a piece of layer n in each of the network entities. These pieces communicate with each other by exchanging layer-n messages. These messages are called layer-n protocol data units, or more commonly n-pdus. The contents and format of an n-pdu, as well as the manner in which the n-pdus are exchanged among the network elements, are defined by a layer-n protocol. When taken together, the protocols of the various layers are called the protocol stack. When layer n of host A sends an n-pdu to layer n of host B, layer n of host A passes the n-pdu to layer n 1 and then lets layer n 1 deliver the n-pdu to layer n of B; thus layer n is said to rely on layer n 1 to deliver its n-pdu to the destination. A key concept is that of the service model of a layer. Layer n 1 is said to offer services to layer n. For example, layer n 1 might guarantee that the n-pdu will arrive without error at layer n in the destination within one second, or it might only guarantee that the n-pdu will eventually arrive at the destination without any assurances about error. Protocol Layering The concept of protocol layering is fairly abstract and is sometimes difficult to grasp at first. This concept will become clear as we study the Internet layers and their constituent protocols in greater detail. But let us now try to shed some insight on protocol layering and protocol stacks with an example. Consider a network that organizes its communication protocols in four layers (which is different from the Internet, which organizes its protocols in five layers), as shown in Figure Because there are four layers, there are four types of PDUs: 1-PDUs, 2-PDUs, 3-PDUs, and 4- PDUs. As shown in Figure 1.22, the application, operating at the highest layer, layer 4, creates a message, M. Any message created at this highest layer is a 4-PDU. The message M itself may consist of many different fields (in much the same way that a structure or record in a programming language may contain different fields); it is up to the application to define and interpret the fields in the message. The fields might contain the name of the sender, a code indicating the type of the message, and some additional data. Within the source host, the contents of the entire message M are then passed down the protocol stack to layer 3. In the example in Figure 1.22, layer 3 in the source host divides a 4-PDU, M, into two parts, M 1 and M 2. The layer 3 in the source host then adds to M 1 and M 2 so-called headers to create two layer-3 PDUs. Headers contain the additional information needed by the sending and receiving

53 C01 pp4 6/14/02 5:45 PM Page Protocol Layers and Their Service Models 53 Original message M M H 3 M 1 H 3 M 2 3-PDU H 3 M 1 H 3 M 2 H 2 H 3 M 1 H 2 H 3 M 2 2-PDU H 2 H 3 M 1 H 2 H 3 M 2 H 1 H 2 H 3 M 1 H 1 H 2 H 3 M 2 1-PDU H 1 H 2 H 3 M 1 H 1 H 2 H 3 M 2 Source Destination Figure 1.22 Different PDUs at different layers in the protocol architecture sides of layer 3 to implement the service that layer 3 provides to layer 4. The procedure continues in the source, adding more header at each layer, until the 1-PDUs are created. The 1-PDUs are sent out of the source host onto a physical link. At the other end, the destination host receives 1-PDUs and directs them up the protocol stack. At each layer, the corresponding header is removed. Finally, M is reassembled from M 1 and M 2 and then passed on to the application. Note that in Figure 1.22, layer n uses the services of layer n 1. For example, once layer 4 creates the message M, it passes the message down to layer 3 and relies on layer 3 to deliver the message to layer 4 at the destination. Interestingly enough, this notion of relying on lower-layer services is prevalent in many other forms of communication. For example, consider ordinary postal mail. When you write a letter, you include envelope information such as the destination address and the return address with the letter. The letter, along with the address information, can be considered a PDU at the highest layer of the protocol stack. You then drop the PDU in a mailbox. At this point, the letter is out of your hands. The postal service may then add some of its own internal information onto your letter, essentially adding a header to your letter. For example, in the United States a barcode is often printed on your letter. Once you drop your envelope into a mailbox, you rely on the services of the postal service to deliver the letter to the correct destination in a timely manner. For example, you don t worry about whether a postal truck will break down while carrying the letter. Instead, the postal service takes care of this, presumably with well-defined plans to recover from such failures. Furthermore, within the postal service itself

54 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet there are layers, and the protocols at one layer rely on and use the services of the layer below. In order for one layer to interoperate with the layer below it, the interfaces between the two layers must be precisely defined. Standards bodies define precisely the interfaces between adjacent layers (for example, the format of the PDUs passed between the layers) and permit the developers of networking software and hardware to implement the interior of the layers as they please. Therefore, if a new and improved implementation of a layer is released, the new implementation can replace the old implementation and, in theory, the layers will continue to interoperate. Layer Functions In a computer network, each layer may perform one or more of the following generic set of tasks: Error control, which makes the logical channel between the layers in two peer network elements more reliable Flow control, which avoids overwhelming a slower peer with PDUs Segmentation and reassembly, which at the transmitting side divides, large data chunk into smaller pieces and at the receiving side reassembles the smaller pieces into the original large chunk Multiplexing, which allows several higher-level sessions to share a single lowerlevel connection Connection setup, which provides handshaking with a peer Protocol layering has conceptual and structural advantages. We mention, however, that some researchers and networking engineers are vehemently opposed to layering [Wakeman 1992]. One potential drawback of layering is that one layer may duplicate lower-layer functionality. For example, many protocol stacks provide error recovery on both a link basis and an end-to-end basis. A second potential drawback is that functionality at one layer may need information (for example, a timestamp value) that is present only in another layer; this violates the goal of separation of layers The Internet Protocol Stack The Internet stack consists of five layers: the physical, data link, network, transport, and application layers. Rather than use the cumbersome terminology n-pdu for each of the five layers, we instead give special names to the PDUs in four of the five layers: frame, datagram, segment, and message. We don t name a data unit for the physical layer, as no name is commonly used at this layer. The Internet stack and the corresponding PDU names are illustrated in Figure A protocol layer can be implemented in software, in hardware, or in a combination of the two. Application-layer protocols such as HTTP and SMTP are almost

55 C01 pp4 6/14/02 5:45 PM Page Protocol Layers and Their Service Models 55 Stack PDUs Layer 5 Application Message Layer 4 Transport Segment Layer 3 Network Datagram Layer 2 Link Frame Layer 1 Physical 1-PDU Figure 1.23 The Internet protocol stack, and protocol data units always implemented in software in the end systems; so are transport-layer protocols. Because the physical layer and data link layers are responsible for handling communication over a specific link, they are typically implemented in a network interface card (for example, Ethernet or ATM interface cards) associated with a given link. The network layer is often a mixed implementation of hardware and software. We now summarize the Internet layers and the services they provide. Application Layer The application layer is responsible for supporting network applications. The application layer includes many protocols, including HTTP to support the Web, SMTP to support electronic mail, and FTP to support file transfer. We ll see in Chapter 2 that it is very easy to create our own new application-layer protocols. Transport Layer The transport layer provides the service of transporting application-layer messages between the client and server sides of an application. In the Internet there are two transport protocols, TCP and UDP, either of which can transport application-layer messages. TCP provides a connection-oriented service to its applications. This service includes guaranteed delivery of application-layer messages to the destination and flow control (that is, sender/receiver speed matching). TCP also segments long messages into shorter segments and provides a congestion control mechanism, so that a source throttles its transmission rate when the network is congested. The UDP protocol provides its applications a connectionless service, which (as we saw in Section 1.2) is very much a no-frills service.

56 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet Network Layer The network layer is responsible for routing datagrams from one host to another. The Internet s network layer has two principal components. It has a protocol that defines the fields in the IP datagram as well as how the end systems and routers act on these fields. This protocol is the celebrated IP protocol. There is only one IP protocol, and all Internet components that have a network layer must run the IP protocol. The Internet s network layer also contains routing protocols that determine the routes that datagrams take between sources and destinations. The Internet has many routing protocols. As we saw in Section 1.3, the Internet is a network of networks, and within a network, the network administrator can run any routing protocol desired. Although the network layer contains both the IP protocol and numerous routing protocols, it is often simply referred to as the IP layer, reflecting the fact that IP is the glue that binds the Internet together. The Internet transport layer protocol (TCP or UDP) in a source host passes a transport-layer segment and a destination address to the IP layer, just as you give the postal service a letter with a destination address. The IP layer then provides the service of delivering the segment to the destination host. When the packet arrives at the destination host, IP passes the segment to the transport layer within the host. Link Layer The network layer routes a packet through a series of packet switches (called routers, in the Internet) between the source and destination. To move a packet from one node (host or packet switch) to the next node in the route, the network layer must rely on the services of the link layer. In particular, at each node, the network layer passes the datagram down to the link layer, which delivers the datagram to the next node along the route. At this next node, the link layer passes the datagram up to the network layer. The services provided by the link layer depend on the specific link-layer protocol that is employed over the link. For example, some protocols provide reliable delivery on a link basis, that is, from transmitting node, over one link, to receiving node. Note that this reliable delivery service is different from the reliable delivery service of TCP, which provides reliable delivery from one end system to another. Examples of link layers include Ethernet and PPP; in some contexts, ATM and frame relay can be considered link layers. As datagrams typically need to traverse several links to travel from source to destination, a datagram may be handled by different link-layer protocols at different links along its route. For example, a datagram may be handled by Ethernet on one link and then PPP on the next link. The network layer will receive a different service from each of the different link-layer protocols. Physical Layer While the job of the link layer is to move entire frames from one network element to an adjacent network element, the job of the physical layer is to move the individual

57 C01 pp4 6/14/02 5:45 PM Page Protocol Layers and Their Service Models 57 bits within the frame from one node to the next. The protocols in this layer are again link dependent and further depend on the actual transmission medium of the link (for example, twisted-pair copper wire, single-mode fiber optics). For example, Ethernet has many physical layer protocols: one for twisted-pair copper wire, another for coaxial cable, another for fiber, and so on. In each case, a bit is moved across the link in a different way. If you examine the Table of Contents, you will see that we have roughly organized this book using the layers of the Internet protocol stack. We take a top-down approach, first covering the application layer and then proceeding downward Network Entities and Layers The most important network entities are end systems and packet switches. As we discuss later in this book, there are two types of packet switches: routers and bridges. We presented an overview of routers in the earlier sections. Bridges will be discussed in detail in Chapter 5, whereas routers will be covered in more detail in Chapter 4. Similar to end systems, routers and bridges organize the networking hardware and software into layers. But routers and bridges do not implement all of the layers in the protocol stack; they typically implement only the bottom layers. As shown in Figure 1.24, bridges implement layers 1 and 2; routers implement layers Source Data Application Transport Network Link Physical Link Physical Network Link Physical Data Bridge Router Destination Application Transport Network Link Physical Figure 1.24 Hosts, routers, and bridges; each contains a different set of layers, reflecting their differences in functionality

58 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet 1 through 3. This means, for example, that Internet routers are capable of implementing the IP protocol (a layer 3 protocol), while bridges are not. We will see later that while bridges do not recognize IP addresses, they are capable of recognizing layer 2 addresses, such as Ethernet addresses. Note that hosts implement all five layers; this is consistent with the view that the Internet architecture puts much of its complexity at the edges of the network. 1.8 History of Computer Networking and the Internet Sections presented an overview of the technology of computer networking and the Internet. You should know enough now to impress your family and friends! However, if you really want to be a big hit at the next cocktail party, you should sprinkle your discourse with tidbits about the fascinating history of the Internet [Segaller 1998] The Development of Packet Switching: The field of computer networking and today s Internet trace their beginnings back to the early 1960s, a time at which the telephone network was the world s dominant communication network. Recall from Section 1.3 that the telephone network uses circuit switching to transmit information from a sender to receiver an appropriate choice given that voice is transmitted at a constant rate between sender and receiver. Given the increasing importance (and great expense) of computers in the early 1960s and the advent of timeshared computers, it was perhaps natural (at least with perfect hindsight!) to consider the question of how to hook computers together so that they could be shared among geographically distributed users. The traffic generated by such users was likely to be bursty intervals of activity, such as the sending of a command to a remote computer, followed by periods of inactivity while waiting for a reply or while contemplating the received response. Three research groups around the world, all unaware of the others work [Leiner 1998], began inventing the notion of packet switching as an efficient and robust alternative to circuit switching. The first published work on packet-switching techniques was that of Leonard Kleinrock [Kleinrock 1961; Kleinrock 1964], at that time a graduate student at MIT. Using queuing theory, Kleinrock s work elegantly demonstrated the effectiveness of the packet-switching approach for bursty traffic sources. In 1964, Paul Baran [Baran 1964] at the Rand Institute had begun investigating the use of packet switching for secure voice over military networks, and at the National Physical Laboratory in England, Donald Davies and Roger Scantlebury were also developing their ideas on packet switching. The work at MIT, Rand, and NPL laid the foundations for today s Internet. But the Internet also has a long history of a let s-build-it-and-demonstrate-it attitude that

59 C01 pp4 6/14/02 5:45 PM Page History of Computer Networking and the Internet 59 Figure 1.25 The first interface message processor (IMP), with L. Kleinrock also dates back to the early 1960s. J.C.R. Licklider [DEC 1990] and Lawrence Roberts, both colleagues of Kleinrock s at MIT, went on to lead the computer science program at the Advanced Research Projects Agency (ARPA) in the United States. Roberts published an overall plan for the so-called ARPAnet [Roberts 1967], the first packet-switched computer network and a direct ancestor of today s public Internet. The early packet switches were known as interface message processors (IMPs), and the contract to build these switches was awarded to the BBN company. On Labor Day in 1969, the first IMP was installed at UCLA under Kleinrock s supervision, with three additional IMPs being installed shortly thereafter at the Stanford Research Institute (SRI), UC Santa Barbara, and the University of Utah (Figure 1.25). The fledgling precursor to the Internet was four nodes large by the end of Kleinrock recalls the very first use of the network to perform a remote login from UCLA to SRI, crashing the system [Kleinrock 1998]. By 1972, ARPAnet had grown to approximately 15 nodes, and was given its first public demonstration by Robert Kahn at the 1972 International Conference on Computer Communications. The first host-to-host protocol between ARPAnet end systems, known as the network-control protocol (NCP), was completed [RFC 001].

60 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet With an end-to-end protocol available, applications could now be written. The first program was written by Ray Tomlinson at BBN in Proprietary Networks and Internetworking: The initial ARPAnet was a single, closed network. In order to communicate with an ARPAnet host, one had to be actually attached to another ARPAnet IMP. In the early to mid-1970s, additional packet-switching networks besides ARPAnet came into being: ALOHAnet, a microwave network linking together universities on the Hawaiian islands [Abramson 1970] Telenet, a BBN commercial packet-switching network based on ARPAnet technology Cyclades, a French packet-switching network pioneered by Louis Pouzin [Think 2002] Time-sharing networks such as Tymnet and the GE Information Services network, among others in the late 1960s to early 1970s [Schwartz 1977] IBM s SNA ( ), which paralleled the ARPAnet work [Schwartz 1977] The number of networks was beginning to grow. In 1973, Robert Metcalfe s Ph.D. thesis laid out the principle of Ethernet, which would later lead to a huge growth in so-called local area networks (LANs) that operated over a small distance based on the Ethernet protocol. Once again, with perfect hindsight one might now see that the time was ripe for developing an encompassing architecture for connecting networks together. Pioneering work on interconnecting networks (once again under the sponsorship of DARPA Defense Advanced Research Projects Agency), in essence creating a network of networks, was done by Vinton Cerf and Robert Kahn [Cerf 1974]; the term internetting was coined to describe this work. These architectural principles were embodied in the TCP protocol. The early versions of TCP, however, were quite different from today s TCPs. The early versions of TCP combined a reliable in-sequence delivery of data via end-system retransmission (still part of today s TCP) with forwarding functions (which today are performed by IP). Early experimentation with TCP, combined with the recognition of the importance of an unreliable, non-flow-controlled end-end transport service for applications such as packetized voice, led to the separation of IP out of TCP and the development of the UDP protocol. The three key Internet protocols that we see today TCP, UDP, and IP were conceptually in place by the end of the 1970s. In addition to the DARPA Internet-related research, many other important networking activities were under way. In Hawaii, Norman Abramson was developing

61 C01 pp4 6/14/02 5:45 PM Page History of Computer Networking and the Internet 61 Figure 1.26 Metcalfe s original conception of the Ethernet ALOHAnet, a packet-based radio network that allowed multiple remote sites on the Hawaiian islands to communicate with each other. The ALOHA protocol [Abramson 1970] was the first so-called multiple-access protocol, allowing geographically distributed users to share a single broadcast communication medium (a radio frequency). Abramson s work on multiple-access protocols was built upon by Metcalfe and Boggs in the development of the Ethernet protocol [Metcalfe 1976] for wirebased shared broadcast networks; see Figure Interestingly, Metcalfe and Boggs Ethernet protocol was motivated by the need to connect multiple PCs, printers, and shared disks together [Perkins 1994]. Twenty-five years ago, well before the PC revolution and the explosion of networks, Metcalfe and Boggs were laying the foundation for today s PC LANs. Ethernet technology represented an important step for internetworking as well. Each Ethernet local area network was itself a network, and as the number of LANs proliferated, the need to internetwork these LANs together became increasingly important. We discuss Ethernet, ALOHA, and other LAN technologies in detail in Chapter A Proliferation of Networks: By the end of the 1970s, approximately 200 hosts were connected to the ARPAnet. By the end of the 1980s the number of hosts connected to the public Internet, a confederation of networks looking much like today s Internet, would reach 100,000. The 1980s would be a time of tremendous growth.

62 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet Much of the growth in the early 1980s resulted from several distinct efforts to create computer networks linking universities together. BITNET provided and file transfers among several universities in the Northeast. CSNET (computer science network) was formed to link together university researchers without access to ARPAnet. In 1986, NSFNET was created to provide access to NSF-sponsored supercomputing centers. Starting with an initial backbone speed of 56 Kbps, NSFNET s backbone would be running at 1.5 Mbps by the end of the decade, and would be serving as a primary backbone linking together regional networks. In the ARPAnet community, many of the final pieces of today s Internet architecture were falling into place. January 1, 1983, saw the official deployment of TCP/IP as the new standard host protocol for ARPAnet (replacing the NCP protocol). The transition [RFC 801] from NCP to TCP/IP was a flag day type event all hosts were required to transfer over to TCP/IP as of that day. In the late 1980s, important extensions were made to TCP to implement host-based congestion control [Jacobson 1988]. The Domain Name System, used to map between a humanreadable Internet name (for example, gaia.cs.umass.edu) and its 32-bit IP address, was also developed [RFC 1034]. Paralleling this development of the ARPAnet (which was for the most part a United States effort), in the early 1980s the French launched the Minitel project, an ambitious plan to bring data networking into everyone s home. Sponsored by the French government, the Minitel system consisted of a public packet-switched network (based on the X.25 protocol suite, which uses virtual circuits), Minitel servers, and inexpensive terminals with built-in low-speed modems. The Minitel became a huge success in 1984 when the French government gave away a free Minitel terminal to each French household that wanted one. Minitel sites included free sites such as a telephone directory site as well as private sites, which collected a usage-based fee from each user. At its peak in the mid 1990s, it offered more than 20,000 different services, ranging from home banking to specialized research databases. It was used by over 20 percent of France s population, generated more than $1 billion in revenue each year, and created 10,000 jobs. The Minitel was in a large proportion of French homes ten years before most Americans had ever heard of the Internet The Internet Explosion: The 1990s The 1990s were ushered in with a number of events that symbolized the continued evolution and the soon-to-arrive commercialization of the Internet. ARPAnet, the progenitor of the Internet, ceased to exist. MILNET and the Defense Data Network had grown in the 1980s to carry most of the U.S. Department of Defense related traffic and NSFNET had begun to serve as a backbone network connecting regional networks in the United States and national networks overseas. In 1991, NSFNET lifted its restrictions on use of NSFNET for commercial purposes. NSFNET itself would be decommissioned in 1995, with Internet backbone traffic being carried by commercial Internet service providers.

63 C01 pp4 6/14/02 5:45 PM Page History of Computer Networking and the Internet 63 The main event of the 1990s, however, was to be the emergence of the World Wide Web, which brought the Internet into the homes and businesses of millions of people worldwide. The Web also served as a platform for enabling and deploying hundreds of new applications, including online stock trading and banking, streamed multimedia services, and information retrieval services. For a brief history of the early days of the Web, see [W3C 1995]. The Web was invented at CERN by Tim Berners-Lee in [Berners- Lee 1989], based on ideas originating in earlier work on hypertext from the 1940s by Bush [Bush 1945] and since the 1960s by Ted Nelson [Ziff-Davis 1998]. Berners- Lee and his associates developed initial versions of HTML, HTTP, a Web server, and a browser the four key components of the Web. The original CERN browsers provided only a line-mode interface. Around the end of 1992 there were about 200 Web servers in operation, this collection of servers being the tip of the iceberg for what was about to come. At about this time several researchers were developing Web browsers with GUI interfaces, including Marc Andreesen, who led the development of the popular GUI browser Mosaic. Andreesen and his colleagues released an alpha version of their browser in 1993, and in 1994 he and Jim Clark formed Mosaic Communications, which later became Netscape Communications Corporation [Cusumano 1998; Quittner 1998]. By 1995, university students were using Mosaic and Netscape browsers to surf the Web on a daily basis. At about this time companies big and small began to operate Web servers and transact commerce over the Web. In 1996, Microsoft started to make browsers, which started the browser war between Netscape and Microsoft, which Microsoft won a few years later [Cusumano 1998]. The second half of the 1990s was a period of tremendous growth and innovation for the Internet, with major corporations and thousands of startups creating Internet products and services. Internet continued to evolve with feature-rich mail readers providing address books, attachments, hot links, and multimedia transport. By the end of the millennium the Internet was supporting hundreds of popular applications, including four killer applications: , including attachments and Web-accessible The Web, including Web browsing and Internet commerce Instant messaging, with contact lists, pioneered by ICQ Peer-to-peer file sharing of MP3s, pioneered by Napster Interestingly, the first two killer applications came from the research community, whereas the last two were created by a few young entrepreneurs. The period was also a roller-coaster ride for the Internet in the financial markets. Before becoming profitable companies, hundreds of Internet startups IPOed (that is, started to be traded in a stock market with an Initial Public Offering). Many companies were valued in the billions of dollars without having

64 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet any significant revenue streams. The Internet stocks collapsed in , with many startups shutting down. Nevertheless, a number of companies emerged as big winners in the Internet space (even if their stock prices suffered in the crash), including Microsoft, Cisco, AOL, and Yahoo. During the 1990s, networking research and development also made significant advances in the areas of high-speed routers and routing (see Chapter 4) and local area networks (see Chapter 5). The technical community struggled with the problems of defining and implementing an Internet service model for traffic requiring real-time constraints, such as continuous media applications (see Chapter 6). The need to secure and manage Internet infrastructure (see Chapters 7 and 8) also became of paramount importance as e-commerce applications proliferated and the Internet became a central component of the world s telecommunications infrastructure Recent Developments Innovation in computer networking continues at a rapid pace. Advances are being made on all fronts, including deployment of new applications, security, content distribution, Internet telephony, higher transmission speeds in LANs, and faster routers. But there are three developments that merit special attention: broadband residential Internet access, wireless Internet access, and peer-to-peer (P2P) networking. Broadband residential Internet access, using cable modem and DSL technology (discussed in Section 1.4), is enjoying rapid deployment throughout the world. Many analysts predict that by 2005 roughly 50 percent of residential access will be broadband. This sets the stage for a wealth of new multimedia applications, including streaming high-quality video-on-demand and high-quality interactive video conferencing. Wireless Internet access has already been deployed in Japan with tremendous success with i-mode, NTT s DoCoMo s mobile Internet access system [DoCoMo 2002]. The typical i-mode handset resembles an ordinary mobile phone but with a somewhat larger screen for displaying text and images. The handsets provide both mobile telephone and mobile Internet access. As of August 2001, there were over 20 million i-mode subscribers in Japan, and the number continues to grow rapidly. In Europe and North America, numerous deployments of wireless Internet technologies are currently taking place. The i-mode technology leverages the mobile phone infrastructure to provide low-speed wide-area mobile access to the Internet. The first few years of the new decade also witnessed the rapid deployment of wireless LANs, providing higher-speed wireless access to the Internet from hotels, cafes, corporate buildings, university buildings, and homes. The last innovation of which we take special note is that of P2P networking. A P2P networking application exploits the resources in users computers storage, content, CPU cycles, and human presence and has significant autonomy

65 C01 pp4 6/14/02 5:45 PM Page Summary 65 from central servers. Typically, the users computers (that is, the peers ) have intermittent connectivity. Napster was the first successful P2P file-sharing application for MP3s. More recent P2P file-sharing applications distribute not only MP3s but also videos, images, and text files. Instant messaging systems with contact lists are P2P communication applications since messages are sent directly between pairs of peers (typically over TCP connections) without passing through central servers. Finally, the successful SETI@home project is an example of P2P computing. 1.9 Summary In this chapter we ve covered a tremendous amount of material! We ve looked at the various pieces of hardware and software that make up the Internet in particular, and computer networks in general. We started at the edge of the network, looking at end systems and applications, and at the transport service provided to the applications running on the end systems. Using network-based distributed applications as examples, we introduced the notion of a protocol a key concept in networking. We then dove deeper inside the network, into the network core, identifying packet switching and circuit switching as the two basic approaches for transporting data through a telecommunication network, and we examined the strengths and weaknesses of each approach. We then looked at the lowest (from an architectural standpoint) parts of the network the link-layer technologies and physical media typically found in the access network. We also examined the structure of the global Internet, learning that the Internet is a network of networks. We saw that the Internet s hierarchical structure, consisting of higher- and lower-tier ISPs, has allowed it to scale to include thousands of networks. In the second part of this introductory chapter, we examined several topics that are central to the field of computer networking. We first examined the causes of delay and packet loss in a packet-switched network. We developed simple quantitative models of transmission, propagation, and queuing delays; we will make extensive use of these delay models in the homework problems throughout this book. Next we examined protocol layering and service models, key architectural principles in networking that we will also refer back to throughout this book. We finished our introduction to networking with a brief history of computer networking. The first chapter in itself constitutes a mini-course in computer networking. So, we have indeed covered a tremendous amount of ground in this first chapter! If you re a bit overwhelmed, don t worry. In the following chapters we will revisit all of these ideas, covering them in much more detail (that s a promise, not a threat!). At this point, we hope you leave this chapter with a still-developing intuition for the pieces that make up a network, a still-developing command of the

66 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet vocabulary of networking (don t be shy about referring back to this chapter), and an ever-growing desire to learn more about networking. That s the task ahead of us for the rest of this book. Road-Mapping This Book Before starting any trip, we should always glance at a road map in order to become familiar with the major roads and junctures that lie ahead. For the trip we are about to embark on, the ultimate destination is a deep understanding of the how, what, and why of computer networks. Our road map is the sequence of chapters of this book: 1. Computer Networks and the Internet 2. Application Layer 3. Transport Layer 4. Network Layer and Routing 5. Link Layer and Local Area Networks 6. Multimedia Networking 7. Security in Computer Networks 8. Network Management Taking a look at this road map, we identify Chapters 2 through 5 as the four core chapters of this book. You should notice that there is one chapter for each of the top four layers of the Internet protocol stack. Further note that our journey will begin at the top of the Internet protocol stack, namely, the application layer, and will work its way downward. The rationale behind this top-down journey is that once we understand the applications, we can then understand the network services needed to support these applications. We can then, in turn, examine the various ways in which such services might be implemented by a network architecture. Covering applications early thus provides motivation for the remainder of the text. The second half of the book Chapters 6 through 8 zooms in on three enormously important (and somewhat independent) topics in modern computer networking. In Chapter 6 (Multimedia Networking), we examine audio and video applications such as Internet phone, video conferencing, and streaming of stored media. We also look at how a packet-switched network can be designed to provide consistent quality of service to audio and video applications. In Chapter 7 (Security in Computer Networks), we first look at the underpinnings of encryption and network security, and then we examine how the basic theory is being applied in a broad range of Internet contexts. The last chapter (Network Management) examines the key issues in network management as well as the primary Internet protocols used for network management.

67 C01 pp4 6/14/02 5:45 PM Page 67 Homework Problems and Questions 67 Homework Problems and Questions Chapter 1 Review Questions Sections What is the difference between a host and an end system? List different types of end systems. Is a Web server an end system? 2. The word protocol is often used in the media in terms of diplomatic relations. Give an example of a diplomatic protocol. 3. What is a client program? What is a server program? Does a server program request and receive services from a client program? 4. What are the two types of services that the Internet provides to its applications? What are some characteristics of each of these services? 5. It has been said that flow control and congestion control are equivalent. Is this true for the Internet s connection-oriented service? Are the objectives of flow control and congestion control the same? 6. Provide a brief, high-level description of how the Internet s connection-oriented service provides reliable transport. 7. What advantage does a circuit-switched network have over a packet-switched network? What advantages does TDM have over FDM in a circuit-switched network? 8. Why is it that packet switching is said to employ statistical multiplexing? Contrast statistical multiplexing with the multiplexing that takes place in TDM. 9. Suppose there is exactly one packet switch between a sending host and a receiving host. The transmission rates between the sending host and the switch and between the switch and the receiving host are R 1 and R 2, respectively. Assuming that the switch uses store-and-forward packet switching, what is the total end-to-end delay to send a packet of length L? (Ignore queuing, propagation delay, and processing delay.) 10. What is meant by connection state information in a virtual circuit network? If in a switch in a VC network, connections are established and torn down at a rate of one connection per millisecond (on average), at what rate does the forwarding table in the switch need to be modified? 11. Suppose you are developing a standard for a new type of network. You need to decide whether your network will use VCs or datagram routing. What are the pros and cons for using VCs? 12. What are the advantages of message segmentation in packet-switched networks? What are the disadvantages? 13. List six different access technologies. Classify each one as residential access, company access, or mobile access. 14. What is the key distinguishing difference between a tier-1 ISP and a tier-2 ISP?

68 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet 15. What is the difference between a POP and a NAP? 16. Is HFC bandwidth dedicated or shared among users? Are collisions possible in a downstream HFC channel? Why or why not? 17. What is the transmission rate of Ethernet LANs? For a given transmission rate, can each user on the LAN continuously transmit at that rate? 18. What are some of the physical media that Ethernet can run over? 19. Dial-up modems, HFC, and ADSL are all used for residential access. For each of these access technologies, provide a range of transmission rates and comment on whether the bandwidth is shared or dedicated. Sections Consider sending a series of packets from a sending host to a receiving host over a fixed route. List the delay components in the end-to-end delay for a single packet. Which of these delays are constant and which are variable? 21. List five tasks that a layer can perform. Is it possible that one (or more) of these tasks could be performed by two (or more) layers? 22. What are the five layers in the Internet protocol stack? What are the principal responsibilities of each of these layers? 23. Which layers in the Internet protocol stack does a router process? Problems 1. Design and describe an application-level protocol to be used between an automatic teller machine and a bank s centralized computer. Your protocol should allow a user s card and password to be verified, the account balance (which is maintained at the centralized computer) to be queried, and an account withdrawal to be made (that is, money disbursed to the user). Your protocol entities should be able to handle the all-too-common case in which there is not enough money in the account to cover the withdrawal. Specify your protocol by listing the messages exchanged and the action taken by the automatic teller machine or the bank s centralized computer on transmission and receipt of messages. Sketch the operation of your protocol for the case of a simple withdrawal with no errors, using a diagram similar to that in Figure 1.2. Explicitly state the assumptions made by your protocol about the underlying end-to-end transport service. 2. Consider an application that transmits data at a steady rate (for example, the sender generates an N-bit unit of data every k time units, where k is small and fixed). Also, when such an application starts, it will continue running for a relatively long period of time. Answer the following questions, briefly justifying your answer:

69 C01 pp4 6/14/02 5:45 PM Page 69 Problems 69 a. Would a packet-switched network or a circuit-switched network be more appropriate for this application? Why? b. Suppose that a packet-switched network is used and the only traffic in this network comes from such applications as described above. Furthermore, assume that the sum of the application data rates is less than the capacities of each and every link. Is some form of congestion control needed? Why? 3. Consider the circuit-switched network in Figure 1.5. Recall that there are n circuits on each link. a. What is the maximum number of simultaneous connections that can be in progress at any one time in this network? b. Suppose that all connections are between the switch in the upper-left-hand corner and the switch in the lower-right-hand corner. What is the maximum number of simultaneous connections that can be in progress? 4. Review the car-caravan analogy in Section 1.6. Again assume a propagation speed of 100 km/hour. a. Suppose the caravan travels 200 km, beginning in front of one toll booth, passing through a second toll booth, and finishing just before a third toll booth. What is the end-to-end delay? b. Repeat (a), now assuming that there are seven cars in the caravan instead of ten. 5. Consider sending a file of F 5 M L bits over a path of Q links. Each link transmits at R bps. The network is lightly loaded so that there are no queuing delays. When a form of packet switching is used, the M L bits are broken up into M packets, each packet with L bits. Propagation delay is negligible. a. Suppose the network is a packet-switched virtual circuit network. Denote the VC set-up time by t s seconds. Suppose the sending layers add a total of h bits of header to each packet. How long does it take to send the file from source to destination? b. Suppose the network is a packet-switched datagram network and a connectionless service is used. Now suppose each packet has 2h bits of header. How long does it take to send the file? c. Repeat (b), but assume message switching is used (that is, 2h bits are added to the message, and the message is not segmented). d. Finally, suppose that the network is a circuit-switched network. Further suppose that the transmission rate of the circuit between source and destination is R bps. Assuming t s set-up time and h bits of header appended to the entire file, how long does it take to send the file? 6. Experiment with the message-segmentation Java applet at the book s Web site. Do the delays in the applet correspond to the delays in the previous question? How do link propagation delays affect the overall end-to-end delay for packet switching (with message segmentation) and for message switching?

70 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet 7. Consider sending a large file of F bits from host A to host B. There are two links (and one switch) between A and B, and the links are uncongested (that is, no queuing delays). Host A segments the file into segments of S bits each and adds 40 bits of header to each segment, forming packets of L S bits. Each link has a transmission rate of R bps. Find the value of S that minimizes the delay of moving the file from host A to host B. Disregard propagation delay. 8. This elementary problem begins to explore propagation delay and transmission delay, two central concepts in data networking. Consider two hosts, A and B, connected by a single link of rate R bps. Suppose that the two hosts are separated by m meters, and suppose the propagation speed along the link is s meters/sec. Host A is to send a packet of size L bits to host B. a. Express the propagation delay, d prop, in terms of m and s. b. Determine the transmission time of the packet, d trans, in terms of L and R. c. Ignoring processing and queuing delays, obtain an expression for the endto-end delay. d. Suppose host A begins to transmit the packet at time t 5 0. At time t 5 d trans, where is the last bit of the packet? e. Suppose d prop is greater than d trans. At time t 5 d trans, where is the first bit of the packet? f. Suppose d prop is less than d trans. At time t 5 d trans, where is the first bit of the packet? g. Suppose s , L bits, and R 5 28 Kbps. Find the distance m so that d prop equals d trans. 9. In this problem we consider sending voice from host A to host B over a packetswitched network (for example, Internet phone). Host A converts analog voice to a digital 64 Kbps bit stream on the fly. Host A then groups the bits into 48- byte packets. There is one link between host A and B; its transmission rate is 1 Mbps and its propagation delay is 2 msec. As soon as host A gathers a packet, it sends it to host B. As soon as host B receives an entire packet, it converts the packet s bits to an analog signal. How much time elapses from the time a bit is created (from the original analog signal at host A) until the bit is decoded (as part of the analog signal at host B)? 10. Suppose users share a 1 Mbps link. Also suppose each user requires 100 Kbps when transmitting, but each user transmits only 10 percent of the time. (See the discussion Packet Switching Versus Circuit Switching in Section 1.3.) a. When circuit switching is used, how many users can be supported? b. For the remainder of this problem, suppose packet switching is used. Find the probability that a given user is transmitting. c. Suppose there are 40 users. Find the probability that at any given time, exactly n users are transmitting simultaneously. (Hint: Use the binomial distribution.)

71 C01 pp4 6/14/02 5:45 PM Page 71 Problems 71 d. Find the probability that there are 11 or more users transmitting simultaneously. 11. Consider the discussion in Section 1.3 under the heading of Packet Switching Versus Circuit Switching, in which an example is provided with a 1 Mbps link. Users are generating data at a rate of 100 Kbps when busy, but are busy generating data only with probability p Suppose that the 1 Mbps link is replaced by a 1 Gbps link. a. What is N, the maximum number of users that can be supported simultaneously under circuit switching? b. Now consider packet switching and a user population of M users. Give a formula (in terms of p, M, N) for the probability that more than N users are sending data. 12. Consider the queuing delay in a router buffer (preceding an outbound link). Suppose all packets are L bits, the transmission rate is R bps, and that N packets simultaneously arrive at the buffer every LN/R seconds. Find the average queuing delay of a packet. (Hint: The queuing delay for the first packet is zero; for the second packet L/R; for the third packet 2L/R. The Nth packet has already been transmitted when the second batch of packets arrives.) 13. Consider the queuing delay in a router buffer. Let I denote traffic intensity; that is, I 5 La/R. Suppose that the queuing delay takes the form IL/R (1 I) for I < 1. a. Provide a formula for the total delay, that is, the queuing delay plus the transmission delay. b. Plot the total delay as a function of L/R. 14. a. Generalize the end-to-end delay formula in Section 1.6 for heterogeneous processing rates, transmission rates, and propagation delays. b. Repeat (a), but now also suppose that there is an average queuing delay of d queue at each node. 15. Perform a Traceroute between source and destination on the same continent at three different hours of the day. a. Find the average and standard deviation of the round-trip delays at each of the three hours. b. Find the number of routers in the path at each of the three hours. Did the paths change during any of the hours? c. Try to identify the number of ISP networks that the Traceroute packets pass through from source to destination. Routers with similar names and/or similar IP addresses should be considered as part of the same ISP. In your experiments, do the largest delays occur at the peering interfaces between adjacent ISPs?

72 C01 pp4 6/14/02 5:45 PM Page CHAPTER 1 Computer Networks and the Internet d. Repeat the above for a source and destination on different continents. Compare the intracontinent and intercontinent results. 16. Suppose two hosts, A and B, are separated by 10,000 kilometers and are connected by a direct link of R 5 1 Mbps. Suppose the propagation speed over the link is meters/sec. a. Calculate the bandwidth-delay product, R t prop. b. Consider sending a file of 400,000 bits from host A to host B. Suppose the file is sent continuously as one big message. What is the maximum number of bits that will be in the link at any given time. c. Provide an interpretation of the bandwidth-delay product. d. What is the width (in meters) of a bit in the link? Is it longer than a football field? e. Derive a general expression for the width of a bit in terms of the propagation speed s, the bandwidth R, and the length of the link m. 17. Referring to problem 16, suppose we can modify R. For what value of R is the width of a bit as long as the length of the link? 18. Consider problem 16 but now with a link of R 5 1 Gbps. a. Calculate the bandwidth-delay product, R t prop. b. Consider sending a file of 400,000 bits from host A to host B. Suppose the file is sent continuously as one big message. What is the maximum number of bits that will be in the link at any given time? c. What is the width (in meters) of a bit in the link? 19. Refer again to problem 16. a. How long does it take to send the file, assuming it is sent continuously? b. Suppose now the file is broken up into ten packets with each packet containing 40,000 bits. Suppose that each packet is acknowledged by the receiver and the transmission time of an acknowledgment packet is negligible. Finally, assume that the sender cannot send a packet until the preceding one is acknowledged. How long does it take to send the file? c. Compare the results from (a) and (b). 20. Suppose there is a 10 Mbps microwave link between a geostationary satellite and its base station on Earth. Every minute the satellite takes a digital photo and sends it to the base station. Assume a propagation speed of meters/sec. a. What is the propagation delay of the link? b. What is the bandwidth-delay product, R t prop? c. Let x denote the size of the photo. What is the minimum value of x for the microwave link to be continuously transmitting? 21. Consider the airline travel analogy in our discussion of layering in Section 1.7, and the addition of headers to protocol data units as they flow down the proto-

73 C01 pp4 6/14/02 5:45 PM Page 73 Discussion Questions 73 col stack. Is there an equivalent notion of header information that is added to passengers and baggage as they move down the airline protocol stack? Discussion Questions 1. Write a one-paragraph description for each of three major projects currently under way at the World Wide Web Consortium (W3C). 2. Using the b wireless LAN technology, design a home network for your home or your parents home. List the specific product models in your home network along with their costs. 3. What is PC-to-phone? Find some of the Web sites of companies that are in the PC-to-phone business. 4. What is Short Message Service (SMS)? Is this service popular in any part of the world, and if so, where and how popular? Is it possible to send an SMS message from a Web site to a portable phone? 5. What is streaming of stored audio? Describe some of the existing products for Internet audio streaming. Find some of the Web sites of companies that are in the Internet audio-streaming business. 6. What is Internet video conferencing? Describe some of the existing products for Internet video conferencing. Find some of the Web sites of companies that are in the Internet video-conferencing business. 7. What is peer-to-peer file sharing? Find five companies that provide peer-topeer file sharing. For each company, what kind of files (that is, content) do they handle? 8. What is instant messaging? Are there any products that allow you to access an instant messaging service through a hand-held device? 9. Who invented ICQ, the first instant messaging service? When was it invented, and how old were the inventors? Similarly, who invented Napster? When was it invented, and how old were the inventors? 10. Surf the Web to find a company that is offering HFC Internet access. What is the transmission rate of the cable modem? Is this rate always guaranteed for each user on the network? 11. Suppose you are developing an application for the Internet. Would you have your application run over TCP or UDP? Elaborate. (We will explore this question in some detail in subsequent chapters. For now, appeal to your intuition to answer the question.)

74 C01 pp4 6/14/02 5:45 PM Page 74 Interview Leonard Kleinrock Leonard Kleinrock is a Professor of Computer Science at the University of California, Los Angeles. In 1969, his computer at UCLA became the first node of the Internet. His creation of packet switching principles in 1961 became the technology behind the Internet. Leonard is also the chairman and founder of Nomadix, Inc., a company whose technology provides greater accessibility of broadband Internet service. He received his BEE from the City College of New York (CCNY) and his Masters and Ph.D. in Electrical Engineering from MIT. What made you decide to specialize in networking/internet technology? As a Ph.D. student at MIT in 1959, I looked around and found that most of my classmates were doing research in the area of information theory and coding theory. At MIT, there was the great researcher Claude Shannon who had launched these fields and had solved most of the important problems already. The research problems that were left were hard and of lesser consequence. So I decided to launch out in a new area that no one else had yet conceived of. Remember that at MIT I was surrounded by lots of computers, and it was clear to me that soon these machines would need to communicate with each other. At the time, there was no effective way for them to do so, so I decided to develop the technology that would permit efficient data networks to be created. What was your first job in the computer industry? What did it entail? I went to the evening session at CCNY from for my bachelor s degree in electrical engineering. During the day, I worked first as a technician and then as an engineer at a small, industrial electronics firm called Photobell. While there, I introduced digital technology to their product line. Essentially, we were using photoelectric devices to detect the presence of certain items (boxes, people, etc.) and the use of a circuit known then as a bistable multivibrator was just the kind of technology we needed to bring digital processing into this field of detection. These circuits happen to be the building blocks for computers, and have come to be known as flip-flops or switches in today s vernacular. 74 What was going through your mind when you sent the first host-to-host message (from UCLA to the Stanford Research Institute)? The first host-to-host message was a bit of an anticlimax. In my mind, the more impressive first event took place on September 2, 1969, when the first piece of networking equipment (the IMP) connected to the first operational system in the outside world (my host computer at UCLA). That s when the Internet was born. Earlier that year, I was quoted in a UCLA press release saying that once the network was up and running, it would be possible to gain access to computer utilities from our

75 C01 pp4 6/14/02 5:45 PM Page 75 homes and offices as easily as we gain access to electricity and telephone connectivity. So my vision at that time was that the Internet would be ubiquitous, always on, always available, anyone with any device could connect from any location, and it would be invisible. However, I never anticipated that my 94-year-old mother would be on the Internet today and indeed she is! What is your vision for the future of networking? The clearest part of my vision is that of nomadic computing and smart spaces. The availability of lightweight, inexpensive, high-performance, portable computing devices plus the ubiquity of the Internet has enabled us to become nomads. Nomadic computing refers to the technology that enables end users who travel from place to place to gain access to Internet services in a transparent fashion, no matter where they travel. However, nomadic computing is only one step. The next step will enable us to move out from the netherworld of cyberspace to the physical world of smart spaces. Our environments (desks, walls, vehicles, watches, belts, and so on) will come alive with technology, through actuators, sensors, logic, processing, storage, cameras, microphones, speakers, displays, and communication. This embedded technology will allow our environment to provide the IP services we want. When I walk into a room, the room will know I entered. I will be able to communicate with my environment naturally, as in spoken English; my requests will generate replies that present Web pages to me from wall displays, through my eyeglasses, as speech, holograms, and so forth. Looking a bit further out, I see a networking future that includes the following additional key components. I see intelligent software agents deployed across the network whose function it is to mine data, act on that data, observe trends, and carry out tasks dynamically and adaptively. I see considerably more network traffic generated not so much by humans, but by these embedded devices and these intelligent software agents. I see large collections of self-organizing systems controlling this vast fast network. I see huge amounts of information flashing across this network instantaneously with this information undergoing enormous processing and filtering. The Internet will essentially be a pervasive global nervous system. I see all these things and more as we move headlong through the 21st century. What people have inspired you professionally? By far, it was Claude Shannon from MIT, a brilliant researcher who had the ability to relate his mathematical ideas to the physical world in highly intuitive ways. He was on my Ph.D. thesis committee. Do you have any advice for students entering the networking/internet field? The Internet and all that it enables is a vast new frontier, full of amazing challenges. There is room for great innovation. Don t be constrained by today s technology. Reach out and imagine what could be and then make it happen. 75